From solko at gcdf.pl Mon Jun 1 01:21:34 2009 From: solko at gcdf.pl (Szymon Olko) Date: Mon, 01 Jun 2009 10:21:34 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <200906010938513832282@163.com> References: <200906010938513832282@163.com> Message-ID: <4A238F8E.8060400@gcdf.pl> zhaoxxqq pisze: > HI, > I use event socket to send command to FS conference. > I send " conference testconf play /root/test.wav" in console. It worked ok. > I send "api conference testconf play /record/test.wav" by event socket. > and the response is"Disconneted, Good bye.See you at ClueCon..". > I changed the wav file to www root. the same problem. can you help me? > 2009-06-01 Do you use 'auth ClueCon' before sending 'api' command? Szymon From codecomplete at free.fr Mon Jun 1 02:11:13 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 1 Jun 2009 02:11:13 -0700 (PDT) Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <1243809075.31679.2.camel@sodium> References: <23807353.post@talk.nabble.com> <1243809075.31679.2.camel@sodium> Message-ID: <23811643.post@talk.nabble.com> Hadley Rich wrote: > The issue you are going to have is that the IP0x are based on the > blackfin processor which as far as I'm aware FreeSWITCH doesn't compile on > yet. Last I heard there's an issue with APR. Too bad. Thanks for the tip. -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p23811643.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 1 02:57:07 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 11:57:07 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. /Peter -----Ursprungligt meddelande----- Fr?n: Peter Olsson Skickat: den 30 maj 2009 09:01 Till: freeswitch-users at lists.freeswitch.org ?mne: FW: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? And just to be clear, even though media flows in one direction (from FS to phone), I get no audio at all. And by the way, I mean SVN, not SNV :) Sorry for double posting... /Peter ________________________________________ Fr?n: Peter Olsson Skickat: den 30 maj 2009 08:58 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Michael Jerris [mike at jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I'm on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: Nope - it's not :) Just to make sure I even deleted the source completely, and checked everything out again. Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682! From anthony.minessale at gmail.com Mon Jun 1 05:59:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 07:59:30 -0500 Subject: [Freeswitch-users] Custom variable and channel answer event (socket) In-Reply-To: <4A232666.70000@gmail.com> References: <4A232666.70000@gmail.com> Message-ID: <191c3a030906010559j5b32ffd6j7c329c9283dcd9df@mail.gmail.com> all the variables should be present in that event. Maybe rebuild to make sure you don't have skew from an upgrade. On Sun, May 31, 2009 at 7:52 PM, paul.degt at gmail.com wrote: > Hi, > I am setting a custom var from a javascript code, I do see it in channel > state events and others up to channel answer. In channel answer event it > somehow disappears, > and then comes back in channel destroy event. My problem is that I > really need it in channel answer event. > What can be wrong here? I did put verbose_event action everywhere I > could think of. > Help would be greatly appreciated. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/a0ad47f9/attachment.html From steveu at coppice.org Mon Jun 1 06:02:22 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 01 Jun 2009 21:02:22 +0800 Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <23807353.post@talk.nabble.com> References: <23807353.post@talk.nabble.com> Message-ID: <4A23D15E.3040908@coppice.org> Fred-145 wrote: > Hello > > Atcom's IP01 unit (www.atcom.cn) can be expanded to have 128MB RAM and 1GB > NAND flash. Before I go ahead and check, would someone know if a minimal > Linux + LAMP server* + Freeswitch can run OK with this amount of memory? > > Thank you. > > * I think I'll trade MySQL with Firebird, to avoid buying a license for my > commercial application > I don't know any free telephony project which will build and run on an IP01, except the ones which have specifically been adapted for the Blackfin. If you look at www.rowetel.com or www.astfin.org you will find versions of Asterisk which have been adapted for the IP01. Nobody has yet adapted Freeswitch for the Blackfin, and they probably won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux, which is a cut down Linux for machines of this type. It is quite troublesome to get memory management to behave sanely on a machine without an MMU. The Asterisk adaptions for the Blackfin have problems with this too, but if you don't let the memory become too fragmented they work OK. The lack of floating point hardware in the Blackfin, and a number of other embedded processors, can also cause performance issues. The core functions of Freeswitch work reasonably well with emulated floating point, but some things, like the FAX engine, are really too slow to be very practical until more of the code is adapted to provide a fixed point version. Steve From anthony.minessale at gmail.com Mon Jun 1 06:08:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 08:08:12 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A21E87E.90709@gmail.com> References: <4A1BFECE.7070603@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> Message-ID: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> I added support so when multiple endconf users are in the same conference the total number of people with the flag must reach 0 before it kills the conf. Can I take a small break now please ;) That's why I'm afraid to add new stuff sometimes. On Sat, May 30, 2009 at 9:16 PM, wrote: > I think I can answer my own question after looking at the code... It > seems that when THAT ONE user leaves, a flag is set that notifies the > conference thread to teardown the conference. I guess I will have to > roll my own on this one I guess, especially since I don't want to kill > the conference completely, just drop the users back to music. > > Also, more importantly... > I just discovered a number of conference profile options that are > neither documented in the Wiki nor mentioned in the sample configuration > file. I've added entries in the Wiki for all the ones that were > missing, but I don't know what half of them do. =( > > Could someone in-the-know please fill those in? Also, I would suggest > adding those to the sample config file. Options like "endconf" and > "announce-user" are GREAT conference features, but no one knows they are > there! > > (I had actually implemented the user count announcement within > Javascript, because I didn't know it was available.) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/351dcf77/attachment.html From kristian.kielhofner at gmail.com Mon Jun 1 06:19:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 1 Jun 2009 09:19:36 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> Message-ID: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson wrote: > Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. > > /Peter > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From larclap at yahoo.com Mon Jun 1 06:20:17 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 1 Jun 2009 06:20:17 -0700 Subject: [Freeswitch-users] Rotating log files not working In-Reply-To: <5a8712120905311837r53efb1d0j3a32ad3661e3bfdf@mail.gmail.com> References: <001801c9e216$287e1800$797a4800$@com> <5a8712120905311837r53efb1d0j3a32ad3661e3bfdf@mail.gmail.com> Message-ID: <004201c9e2bb$b5064ae0$1f12e0a0$@com> I am using version 13441 on Centos 5. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Sunday, May 31, 2009 6:38 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Rotating log files not working Just for the record, always update do latest trunk when testing and provide revision number (version command). Later, jmesquita On Sun, May 31, 2009 at 2:35 PM, Lars Zeb wrote: I am trying to rotate the logs, specifically the cdr ones. But the existing extension and Master csv files are not rotated; they remain untouched. I issue the command ?kill ?s HUP pid? (pid of freeswitch). The fs console says 2009-05-31 10:25:58 [NOTICE] mod_logfile_c:157 mod_logfile_rotate() New log started. The conf/autoload_configs/cdr-csv.conf.xml shows: What am I doing wrong here? Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/32a271ea/attachment-0001.html From peter.olsson at visionutveckling.se Mon Jun 1 06:32:38 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 15:32:38 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Thanks for the reply! Will this really help though? From what I understand of the change that breaks the compatibility, it will always send this header in the "200 OK" message (my problem is incoming calls to FS). The fix you're describing, isn't it when calling from FS to the other end? This way it works either way, it's just when the PBX gets this in the 200 OK message with this header that it stops working. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner Skickat: den 1 juni 2009 15:20 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson wrote: > Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. > > /Peter > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a23d69e32932135138353! From brian at freeswitch.org Mon Jun 1 06:32:59 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:32:59 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> No he's talking about the one from SFSIP-111, Again that shouldn't matter... I'll fix that. /b On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote: > Peter, > > Ouch. Your PBX is broken. It shouldn't do that. > > Luckily FreeSWITCH provides a way to select RPID/PAI/none: > > http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From brian at freeswitch.org Mon Jun 1 06:36:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:36:51 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Message-ID: I'll fix it where it won't do it on anything but polycom. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: > Thanks for the reply! > > Will this really help though? From what I understand of the change > that breaks the compatibility, it will always send this header in > the "200 OK" message (my problem is incoming calls to FS). The fix > you're describing, isn't it when calling from FS to the other end? > This way it works either way, it's just when the PBX gets this in > the 200 OK message with this header that it stops working. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/db9e7759/attachment.html From brian at freeswitch.org Mon Jun 1 06:46:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:46:39 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Message-ID: <24AB9CC7-9E06-44DF-95EB-5437BFFF4324@freeswitch.org> Your PBX is broken but now I have fixed it to only do that feature with Polycom for now. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: > Thanks for the reply! > > Will this really help though? From what I understand of the change > that breaks the compatibility, it will always send this header in > the "200 OK" message (my problem is incoming calls to FS). The fix > you're describing, isn't it when calling from FS to the other end? > This way it works either way, it's just when the PBX gets this in > the 200 OK message with this header that it stops working. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3e1f4fc1/attachment.html From kristian.kielhofner at gmail.com Mon Jun 1 06:51:37 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 1 Jun 2009 09:51:37 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> Message-ID: <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Ahh... That's what I get for not reading the entire thread! On Mon, Jun 1, 2009 at 9:32 AM, Brian West wrote: > No he's talking about the one from SFSIP-111, Again that shouldn't > matter... I'll fix that. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Jun 1 07:02:26 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 09:02:26 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Message-ID: I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote: > Ahh... That's what I get for not reading the entire thread! > > On Mon, Jun 1, 2009 at 9:32 AM, Brian West > wrote: >> No he's talking about the one from SFSIP-111, Again that shouldn't >> matter... I'll fix that. >> >> /b >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From peter.olsson at visionutveckling.se Mon Jun 1 08:08:53 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 17:08:53 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DDB@cooper> Brian - you're the man! :) /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 1 juni 2009 16:02 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote: > Ahh... That's what I get for not reading the entire thread! > > On Mon, Jun 1, 2009 at 9:32 AM, Brian West > wrote: >> No he's talking about the one from SFSIP-111, Again that shouldn't >> matter... I'll fix that. >> >> /b >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a23e16232931225569523! From brian at freeswitch.org Mon Jun 1 08:13:50 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 10:13:50 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <34D78532-AB64-4D28-A6DB-2586390D271A@freeswitch.org> Makes me wanna take a clue-by-4 to some of these devices and just beat them to death! :P /b On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote: > Peter, > > Ouch. Your PBX is broken. It shouldn't do that. > > Luckily FreeSWITCH provides a way to select RPID/PAI/none: > > http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/766c3c34/attachment.html From brian at freeswitch.org Mon Jun 1 08:26:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 10:26:42 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <4A238F8E.8060400@gcdf.pl> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: Did you happen to check out the ClueCon website? Link below! :P /b On Jun 1, 2009, at 3:21 AM, Szymon Olko wrote: > Do you use 'auth ClueCon' before sending 'api' command? > > Szymon Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/832d6d00/attachment.html From asannucci at gmail.com Mon Jun 1 09:39:32 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 18:39:32 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: I Love FreeSWITCH and ClueCon '09 :) The spanish comunity too. www.freeswitch.es -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/7153f99a/attachment.html From brian at freeswitch.org Mon Jun 1 09:49:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 11:49:51 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: So we can look forward to seeing you at ClueCon? /b On Jun 1, 2009, at 11:39 AM, bakko wrote: > I Love FreeSWITCH and ClueCon '09 > > :) > > The spanish comunity too. > www.freeswitch.es > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/efeb9629/attachment.html From fvillarroel at yahoo.com Mon Jun 1 09:54:09 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 09:54:09 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <458378.32320.qm@web34303.mail.mud.yahoo.com> Dear all. I have problem with g729 passthru mode. I received traffic from a Asterisk on my FS and forward to other Asterisk, when i use codec ulaw this works very well. But when i try use G729 i received the following messages and SIP Trace: 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/admin/42452904 at 190.208.xx.yy [f65514e0-4ec7-11de-9b78-150e2985561f] 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/42452904 at 190.208.xx.yy entering state [received][100] 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 25643 25643 IN IP4 190.208.xx.yy s=session c=IN IP4 190.208.xx.yy t=0 0 m=audio 10236 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/42452904 at 190.208.xx.yy [KILL] 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904 at 190.208.xx.yy [BREAK] send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 From: "102" ;tag=bmke36jc1v To: "102" ;tag=rZp0XXrK9NHFD Call-ID: 3c26700b249f-sryanqz0td8u at snom360-00041323143F CSeq: 27056 REGISTER Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 Date: Mon, 01 Jun 2009 16:19:20 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running State Change CS_HANGUP 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/42452904 at 190.208.xx.yy hanging up, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 488 send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 From: "452904" ;tag=as4e2616ae To: ;tag=S8FSZr9p6y71r Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/42452904 at 190.208.xx.yy Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_HANGUP -> CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904 at 190.208.xx.yy [BREAK] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running State Change CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: ------------------------------------------------------------------------ ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport From: "452904" ;tag=as4e2616ae To: ;tag=S8FSZr9p6y71r Contact: Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/42452904 at 190.208.xx.yy Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_REPORTING -> CS_DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) Locked, Waiting on external entities 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) Ended 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/42452904 at 190.208.xx.yy SOFIA DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/42452904 at 190.208.xx.yy Standard DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY going to sleep My vars.xml : I hope your comments for know where is the config problem Fernando. From asannucci at gmail.com Mon Jun 1 09:54:13 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 18:54:13 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: Maybe next year. On August i have to work :(. If you organize ClueCon 2010 on setpember I wil go. :) Good luck. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/97e09d41/attachment.html From msc at freeswitch.org Mon Jun 1 09:59:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 09:59:34 -0700 Subject: [Freeswitch-users] Custom variable and channel answer event (socket) In-Reply-To: <4A232666.70000@gmail.com> References: <4A232666.70000@gmail.com> Message-ID: <87f2f3b90906010959p787a83f1nd051ea3ac460e8a3@mail.gmail.com> Can you pastebin a simple script that demonstrates the issue? Also, include any dialplan/configuration changes you use. Finally, please paste in a sample event with and without the variable in question. Once we have more information we will see what we can do to help. -MC On Sun, May 31, 2009 at 5:52 PM, paul.degt at gmail.com wrote: > Hi, > I am setting a custom var from a javascript code, I do see it in channel > state events and others up to channel answer. In channel answer event it > somehow disappears, > and then comes back in channel destroy event. My problem is that I > really need it in channel answer event. > What can be wrong here? I did put verbose_event action everywhere I > could think of. > Help would be greatly appreciated. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/a5ba4597/attachment.html From stevecrozz at gmail.com Mon Jun 1 10:09:47 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 1 Jun 2009 10:09:47 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> Message-ID: <11990ade0906011009v227a0d80r5b965a48e67ee5dd@mail.gmail.com> No breaks! keep improving the conference app :) --Stephen On Mon, Jun 1, 2009 at 6:08 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I added support so when multiple endconf users are in the same conference > the total number of people with the flag must reach > 0 before it kills the conf. > > Can I take a small break now please ;) > That's why I'm afraid to add new stuff sometimes. > > > > On Sat, May 30, 2009 at 9:16 PM, wrote: > >> I think I can answer my own question after looking at the code... It >> seems that when THAT ONE user leaves, a flag is set that notifies the >> conference thread to teardown the conference. I guess I will have to >> roll my own on this one I guess, especially since I don't want to kill >> the conference completely, just drop the users back to music. >> >> Also, more importantly... >> I just discovered a number of conference profile options that are >> neither documented in the Wiki nor mentioned in the sample configuration >> file. I've added entries in the Wiki for all the ones that were >> missing, but I don't know what half of them do. =( >> >> Could someone in-the-know please fill those in? Also, I would suggest >> adding those to the sample config file. Options like "endconf" and >> "announce-user" are GREAT conference features, but no one knows they are >> there! >> >> (I had actually implemented the user count announcement within >> Javascript, because I didn't know it was available.) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/399de241/attachment.html From anthony.minessale at gmail.com Mon Jun 1 10:11:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 12:11:25 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> It's always august =D it's ok. it counts as work! On Mon, Jun 1, 2009 at 11:54 AM, bakko wrote: > Maybe next year. > > On August i have to work :(. > > If you organize ClueCon 2010 on setpember I wil go. > > :) > > Good luck. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/aa87e413/attachment.html From msc at freeswitch.org Mon Jun 1 10:15:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 10:15:18 -0700 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <458378.32320.qm@web34303.mail.mud.yahoo.com> References: <458378.32320.qm@web34303.mail.mud.yahoo.com> Message-ID: <87f2f3b90906011015u16cf2c0h54829fe8f73372c0@mail.gmail.com> What does your dialplan look like? Just curious where/how you set proxy-media mode. -MC On Mon, Jun 1, 2009 at 9:54 AM, FERNANDO VILLARROEL wrote: > > Dear all. > > I have problem with g729 passthru mode. > > I received traffic from a Asterisk on my FS and forward to other Asterisk, > when i use codec ulaw this works very well. > > But when i try use G729 i received the following messages and SIP Trace: > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel sofia/admin/42452904 at 190.208.xx.yy[f65514e0-4ec7-11de-9b78-150e2985561f] > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/admin/42452904 at 190.208.xx.yy entering state [received][100] > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=root 25643 25643 IN IP4 190.208.xx.yy > s=session > c=IN IP4 190.208.xx.yy > t=0 0 > m=audio 10236 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() > Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() > Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup > sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904 at 190.208.xx.yy [KILL] > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.124:2051 > ;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > From: "102" ;tag=bmke36jc1v > To: "102" ;tag=rZp0XXrK9NHFD > Call-ID: 3c26700b249f-sryanqz0td8u at snom360-00041323143F > CSeq: 27056 REGISTER > Contact: ;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > Date: Mon, 01 Jun 2009 16:19:20 GMT > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running > State Change CS_HANGUP > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > HANGUP > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/admin/42452904 at 190.208.xx.yy hanging up, cause: > INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to > INVITE with: 488 > send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: > ------------------------------------------------------------------------ > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > From: "452904" ;tag=as4e2616ae > To: ;tag=S8FSZr9p6y71r > Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/admin/42452904 at 190.208.xx.yyStandard HANGUP, cause: INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > HANGUP going to sleep > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > Change CS_HANGUP -> CS_REPORTING > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running > State Change CS_REPORTING > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) > State REPORTING > recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: > ------------------------------------------------------------------------ > ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 > Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > From: "452904" ;tag=as4e2616ae > To: ;tag=S8FSZr9p6y71r > Contact: > Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/admin/42452904 at 190.208.xx.yyStandard REPORTING, cause: INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) > State REPORTING going to sleep > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > Change CS_REPORTING -> CS_DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) > Locked, Waiting on external entities > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) > Ended > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) > State DESTROY > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/admin/42452904 at 190.208.xx.yy SOFIA DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/admin/42452904 at 190.208.xx.yyStandard DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) > State DESTROY going to sleep > > My vars.xml : > > data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > > > > I hope your comments for know where is the config problem > > Fernando. > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/41ebc073/attachment-0001.html From msc at freeswitch.org Mon Jun 1 10:16:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 10:16:26 -0700 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> Message-ID: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It's always august =D > it's ok. it counts as work! > If you need a letter to your boss we can do that on ClueCon letterhead. :) -MC > > > > On Mon, Jun 1, 2009 at 11:54 AM, bakko wrote: > >> Maybe next year. >> >> On August i have to work :(. >> >> If you organize ClueCon 2010 on setpember I wil go. >> >> :) >> >> Good luck. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3c59f9a9/attachment.html From intralanman at freeswitch.org Mon Jun 1 10:19:53 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 01 Jun 2009 13:19:53 -0400 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> Message-ID: <4A240DB9.3080208@freeswitch.org> Michael Collins wrote: > > > On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale > > wrote: > > It's always august =D > it's ok. it counts as work! > > > If you need a letter to your boss we can do that on ClueCon letterhead. :) > -MC > my boss paid me for the week last year when i went... any self-respecting employer can't see anything wrong with "Continuing Educational Opportunities" -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3115ff57/attachment.html From asannucci at gmail.com Mon Jun 1 10:29:09 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 19:29:09 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl><191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> Message-ID: <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> I need 5 letters: - one for my boss - one for the boss of my boss - one for my wife - one for my bank (asking more credit) - one for me (like a post-it for don't forget the appointment) If you can do all this, I'will go :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/01d78450/attachment.html From intralanman at freeswitch.org Mon Jun 1 10:33:11 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 01 Jun 2009 13:33:11 -0400 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl><191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> Message-ID: <4A2410D7.8020309@freeswitch.org> > - one for my wife bring her too > - one for my bank (asking more credit) we can write it... but i won't guarantee that they give you what we ask :-P -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/6d012261/attachment.html From nik.middleton at noblesolutions.co.uk Mon Jun 1 12:33:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 20:33:37 +0100 Subject: [Freeswitch-users] Make current fails Message-ID: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Regards, making all mod_sndfile cd . && /bin/sh /usr/src/trunk/libs/libsndfile/missing --run automake-1.9 --gnu configure.ac: required file `Cfg/install-sh' not found configure.ac: required file `Cfg/missing' not found examples/Makefile.am: required file `Cfg/depcomp' not found programs/Makefile.am: required file `Cfg/compile' not found configure.ac:12: required file `Cfg/config.guess' not found configure.ac:12: required file `Cfg/config.sub' not found configure.ac:49: required file `Cfg/ltmain.sh' not found make[6]: *** [Makefile.in] Error 1 make[5]: *** [../../../../libs/libsndfile/src/libsndfile.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_sndfile-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/836db3e9/attachment.html From brian at freeswitch.org Mon Jun 1 12:36:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 14:36:21 -0500 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: > Hi Guys, > > Been running ?make current? and appropriate intervals over the last > few months and all?s been well until today > > Now I get the following, obviously mod_sndfile isn?t happy, but I?m > not sure what to do to fix it > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/92d8d4b6/attachment-0001.html From e at musinghalfwit.org Mon Jun 1 13:12:06 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Mon, 1 Jun 2009 15:12:06 -0500 Subject: [Freeswitch-users] 300 Multiple Choices Message-ID: <20090601201206.GA6838@pointone.com> Hey all, Was curious if there is any 300 Multiple Choices support in freeswitch. It seems from the sofia logging that the sofia library just kills the call when it receives a 300 and freeswitch the same. Is this the case or am I missing something? I googled around and searched the list but didn't see anything definitive. So I just wanted to ensure I hadn't overlooked anything. -Eric From nik.middleton at noblesolutions.co.uk Mon Jun 1 13:52:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 21:52:14 +0100 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Thanks for that ./ bootstrap.sh ./configure Did the trick Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 20:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/84fcfa20/attachment.html From nik.middleton at noblesolutions.co.uk Mon Jun 1 14:18:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 22:18:28 +0100 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Spoke too soon. Clean compile and install, but now FS hangs for about 5 mins on startup Error [unterminated ${var}] in file /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm l line 12 Error including /usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide character) The first error is a typo in the sample, but the second error, I don't have that DIR at all. I presume that this dir has been added, but how to I create these without overwriting my working configs? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 01 June 2009 21:52 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Thanks for that ./ bootstrap.sh ./configure Did the trick Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 20:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/aabf16b3/attachment-0001.html From brian at freeswitch.org Mon Jun 1 14:29:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 16:29:15 -0500 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> I can tell you how to fix it but it'll cost ya! :P /b > Spoke too soon. > > Clean compile and install, but now FS hangs for about 5 mins on > startup > > Error [unterminated ${var}] in file /usr/local/freeswitch/conf/ > autoload_configs/../jingle_profiles/client.xml line 12 > Error including /usr/local/freeswitch/conf/autoload_configs/../ > mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide > character) > > The first error is a typo in the sample, but the second error, I > don?t have that DIR at all. I presume that this dir has been added, > but how to I create these without overwriting my working configs? > > > Regards > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/f12eef55/attachment.html From fvillarroel at yahoo.com Mon Jun 1 15:20:27 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 15:20:27 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <897953.58731.qm@web34301.mail.mud.yahoo.com> Hello the dial plan: This i setup from Wikipbx. --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 2:15 PM > What does your dialplan look like? Just > curious where/how you set proxy-media mode. > -MC > > On Mon, Jun 1, 2009 at 9:54 AM, > FERNANDO VILLARROEL > wrote: > > > > Dear all. > > > > I have problem with g729 passthru mode. > > > > I received traffic from a Asterisk on my FS and forward to > other Asterisk, when i use codec ulaw this works very well. > > > > But when i try use G729 i received the following messages > and SIP Trace: > > > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel > sofia/admin/42452904 at 190.208.xx.yy > [f65514e0-4ec7-11de-9b78-150e2985561f] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 > sofia_handle_sip_i_state() Channel > sofia/admin/42452904 at 190.208.xx.yy entering state > [received][100] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 > sofia_handle_sip_i_state() Remote SDP: > > v=0 > > o=root 25643 25643 IN IP4 190.208.xx.yy > > s=session > > c=IN IP4 190.208.xx.yy > > t=0 0 > > m=audio 10236 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [G729:18:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 > sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 > sofia_handle_sip_i_state() Hangup > sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904 at 190.208.xx.yy [KILL] > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > > send 886 bytes to udp/[190.47.91.83]:60245 at > 16:19:20.596633: > > ? > ------------------------------------------------------------------------ > > ? SIP/2.0 200 OK > > ? Via: SIP/2.0/UDP > 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > > ? From: "102" > ;tag=bmke36jc1v > > ? To: "102" > ;tag=rZp0XXrK9NHFD > > ? Call-ID: > 3c26700b249f-sryanqz0td8u at snom360-00041323143F > > ? CSeq: 27056 REGISTER > > ? Contact: > ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > > > ? Date: Mon, 01 Jun 2009 16:19:20 GMT > > ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > ? Supported: timer, precondition, path, replaces > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) Running State Change > CS_HANGUP > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 > sofia_on_hangup() Channel sofia/admin/42452904 at 190.208.xx.yy > hanging up, cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 > sofia_on_hangup() Responding to INVITE with: 488 > > send 634 bytes to udp/[190.208.xx.yy]:5060 at > 16:19:20.603208: > > ? > ------------------------------------------------------------------------ > > ? SIP/2.0 488 Not Acceptable Here > > ? Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > > ? From: "452904" > ;tag=as4e2616ae > > ? To: > ;tag=S8FSZr9p6y71r > > ? Call-ID: > 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > > ? CSeq: 102 INVITE > > ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > ? Accept: application/sdp > > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > ? Supported: timer, precondition, path, replaces > > ? Allow-Events: talk, refer > > ? Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() > sofia/admin/42452904 at 190.208.xx.yy Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP going to > sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_HANGUP > -> CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) Running State Change > CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING > > recv 408 bytes from udp/[190.208.xx.yy]:5060 at > 16:19:20.620338: > > ? > ------------------------------------------------------------------------ > > ? ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 > > ? Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > > ? From: "452904" > ;tag=as4e2616ae > > ? To: > ;tag=S8FSZr9p6y71r > > ? Contact: > > ? Call-ID: > 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > > ? CSeq: 102 ACK > > ? User-Agent: Asterisk PBX > > ? Max-Forwards: 70 > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() > sofia/admin/42452904 at 190.208.xx.yy Standard REPORTING, > cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING going > to sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State Change > CS_REPORTING -> CS_DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 > (sofia/admin/42452904 at 190.208.xx.yy) Locked, Waiting on > external entities > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 > (sofia/admin/42452904 at 190.208.xx.yy) Ended > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 > sofia_on_destroy() sofia/admin/42452904 at 190.208.xx.yy SOFIA > DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() > sofia/admin/42452904 at 190.208.xx.yy Standard DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY going to > sleep > > > > My vars.xml : > > > > ? data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > ? data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > ? data="xmpp_client_profile=xmppc"/> > > ? data="xmpp_server_profile=xmpps"/> > > > > > > I hope your comments for know where is the config problem > > > > Fernando. > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Mon Jun 1 15:24:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 23:24:52 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> Message-ID: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 22:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails I can tell you how to fix it but it'll cost ya! :P /b Spoke too soon. Clean compile and install, but now FS hangs for about 5 mins on startup Error [unterminated ${var}] in file /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm l line 12 Error including /usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide character) The first error is a typo in the sample, but the second error, I don't have that DIR at all. I presume that this dir has been added, but how to I create these without overwriting my working configs? Regards Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/1434dd7c/attachment-0001.html From brian at freeswitch.org Mon Jun 1 15:33:01 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 17:33:01 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> Message-ID: <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: > Well I can only assume build 13537 is brain dead. Surely I > shouldn?t have to edit a whole bunch of configs to get it working. > FS now takes 3 minutes to start, with no indication as to what it?s > looking for in the logs. That said, to date ?make current? has > always worked well for me. Guess I was bound to hit a bad one > eventually. > > Still, it?s very frustrating. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/4790e407/attachment.html From msc at freeswitch.org Mon Jun 1 15:41:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 15:41:03 -0700 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <897953.58731.qm@web34301.mail.mud.yahoo.com> References: <897953.58731.qm@web34301.mail.mud.yahoo.com> Message-ID: <87f2f3b90906011541x6b27db62ra9231f2b2d9e92f9@mail.gmail.com> On Mon, Jun 1, 2009 at 3:20 PM, FERNANDO VILLARROEL wrote: > > Hello the dial plan: > > > > This i setup from Wikipbx. What about this in the dialplan? Or alternatively this in the SIP profile? I just want to make sure you're actually telling FS to use proxy media. If I may make a suggestion: use pastebin.freeswitch.org and pastebin the entire extension in the dialplan as well as a complete debug log of the call from the FS CLI. Please see this page for some handy tips on gathering information for troubleshooting: http://wiki.freeswitch.org/wiki/Reporting_Bugs -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/fc806580/attachment.html From fvillarroel at yahoo.com Mon Jun 1 16:09:19 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 16:09:19 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <738513.5183.qm@web34308.mail.mud.yahoo.com> Hello i was try with: This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL > wrote: > > > > Hello the dial plan: > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fvillarroel at yahoo.com Mon Jun 1 16:10:06 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 16:10:06 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <592820.20616.qm@web34305.mail.mud.yahoo.com> Hello i was try with: This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL > wrote: > > > > Hello the dial plan: > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brad.tuan at gmail.com Mon Jun 1 20:02:42 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 11:02:42 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: This question can be separated into two part: 1.Pass a call to another FS 2.Receive a call from another FS Somebody can tell me how to do these?? Please............. -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/70a2f185/attachment.html From jcromes at gmail.com Mon Jun 1 20:22:05 2009 From: jcromes at gmail.com (j3flight) Date: Mon, 1 Jun 2009 20:22:05 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> Message-ID: <23825864.post@talk.nabble.com> You rock! Thanks for doing that, makes things work nicely. Although, I was in the process of putting together a php event socket thingy to handle conference state changes. I didn't get very far yet, but it was a good exercise! I do appreciate your work - break granted. Jason -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23825864.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Jun 1 20:23:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 22:23:03 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <191c3a030906012023l7f7dd608k33648dac8f52119e@mail.gmail.com> Try using a telephone? On Mon, Jun 1, 2009 at 10:02 PM, Brad Tuan wrote: > This question can be separated into two part: > 1.Pass a call to another FS > > 2.Receive a call from another FS > Somebody can tell me how to do these?? > > Please............. > > -Brad > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/399edc79/attachment-0001.html From jason at jasonjgw.net Mon Jun 1 20:27:04 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 13:27:04 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602032704.GA5858@jdc.jasonjgw.net> Brad Tuan wrote: > This question can be separated into two part: > 1.Pass a call to another FS uuid_deflect or uuid_transfer, depending on whether the call has been answered by the first FS instance or not. See the wiki. > > 2.Receive a call from another FS Provide a dial plan entry in the second FS that handles the call appropriately after the deflect or transfer. You haven't explained what you're trying to do - a general question warrants a general answer. I am assuming the call is arriving at one FS system, and (before or after answering it), you want to move it to another FS system. That's the question I've answered above. The wiki documents the syntax. From b_ball_henry at hotmail.com Mon Jun 1 20:46:54 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Mon, 1 Jun 2009 20:46:54 -0700 Subject: [Freeswitch-users] Polycom Phone issue Message-ID: <59ad9ca10906012046q48db6b86v69889aafff8ef0ab@mail.gmail.com> I have setup 2 Freeswitch test server, 2 IP430 polycom phones with 2 lines each registered to different server. When I dial extension from the first line (first FS server), if the called party hit reject on the phone. The called won't be hanged up will keep ringing until timed out. When dialed from second line (2nd FS server), the other phone would ring on the screen as of the first line is ringing and besides the caller-id and caller-id-name, it will also show the sip: address of the caller. Now it has similar sympton to the above senario, but more over, even if leg B picks up the call then hang up, Leg A (caller's end) will not hang up for as long as you don't hang it up. Do any of you people with polycom phone have problem like this ? or know what could be the cause? my FS1 server is version 12242M , FS2 server version is 13523M *Here is my dialplan example of the extension that I called:* * And here is the setting for register 2 server on polycom phone:* reg.1.displayName="2025" reg.1.address="2025" reg.1.label="2025" reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="" reg.1.auth.userId="2025" reg.1.auth.password="somepassword" reg.1.server.1.address="10.48.5.83" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.expires.overlap="" reg.1.server.1.register="1" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.outboundProxy.address="" reg.1.outboundProxy.port="" reg.1.outboundProxy.transport="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.proxyRequire="" reg.1.ringType="12" reg.1.lineKeys="1" reg.1.callsPerLineKey="" reg.2.displayName="2025" reg.2.address="2025" reg.2.label="2025" reg.2.type="private" reg.2.lcs="" reg.2.thirdPartyName="" reg.2.auth.userId="2025" reg.2.auth.password="somepassword" reg.2.server.1.address="10.55.7.36" reg.2.server.1.port="5060" reg.2.server.1.transport="DNSnaptr" reg.2.server.2.transport="DNSnaptr" reg.2.server.1.expires="" reg.2.server.1.expires.overlap="" reg.2.server.1.register="" reg.2.server.1.retryTimeOut="" reg.2.server.1.retryMaxCount="" reg.2.server.1.expires.lineSeize="" reg.2.outboundProxy.address="" reg.2.outboundProxy.port="" reg.2.outboundProxy.transport="" reg.2.acd-login-logout="0" reg.2.acd-agent-available="0" -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/47d30482/attachment.html From brad.tuan at gmail.com Mon Jun 1 20:47:54 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 11:47:54 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: >>* This question can be separated into two part: *>>* 1.Pass a call to another FS * >uuid_deflect or uuid_transfer, depending on whether the call has been answered >by the first FS instance or not. See the wiki. >>* *>>* 2.Receive a call from another FS * >Provide a dial plan entry in the second FS that handles the call appropriately >after the deflect or transfer. >You haven't explained what you're trying to do - a general question warrants a >general answer. I am assuming the call is arriving at one FS system, and >(before or after answering it), you want to move it to another FS system. >That's the question I've answered above. The wiki documents the syntax. SorrySorry,let me explain my question. When User1( User of FS1 ) call User2( User of FS2 ) , FS1 will pass the call to FS2 before answering, and then User1 can talk with User2. -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/87cf3686/attachment.html From jason at jasonjgw.net Mon Jun 1 21:04:38 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 14:04:38 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602040438.GA10771@jdc.jasonjgw.net> Brad Tuan wrote: > When User1( User of FS1 ) call User2( User of FS2 ) , > > FS1 will pass the call to FS2 before answering, You just need to write a dial plan extension that matches the call on FS1 and bridges it to FS2. For example: Dialing any extension with the prefix 014 will call that extension (with the prefix removed) on fs2.example.org at port 5080. If FS2 has the default configuration installed, the call will land in the public context of fs2, where you can transfer it to the default context or take other actions depending on the extension called. From brad.tuan at gmail.com Mon Jun 1 22:35:51 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 13:35:51 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: I have tried But the console moniter return : [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187:5080 at external [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED I am sure that 97710001 is a online user on "192.168.141.187", What's wrong?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/412e6cfb/attachment.html From brian at freeswitch.org Mon Jun 1 22:44:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 00:44:56 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <4C970DD9-01C1-4781-9E9B-F67C8401F769@freeswitch.org> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings If you're trying to call a registered user that is NOT the way to do it. sofia/profile/user%domain /b On Jun 2, 2009, at 12:35 AM, Brad Tuan wrote: > I have tried > > > > > > > > But the console moniter return : > [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001- > >01497710001 in context default > [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate > registered user 97710001 at 192.168.141.187:5080 at external > [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A > [CS_NEW] > [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot > create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. > Cause: USER_NOT_REGISTERED > > I am sure that 97710001 is a online user on "192.168.141.187", > What's wrong?? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/7d7357e6/attachment-0001.html From jason at jasonjgw.net Mon Jun 1 22:58:29 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 15:58:29 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602055829.GA29143@jdc.jasonjgw.net> Brad Tuan wrote: > I have tried > > > > > > > > But the console moniter return : > [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 > in context default > [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered > user 97710001 at 192.168.141.187:5080 at external > [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A > [CS_NEW] > [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create > outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: > USER_NOT_REGISTERED > > I am sure that 97710001 is a online user on "192.168.141.187", What's > wrong?? It will work if you use a domain name for the host rather than an IP address. From jason at jasonjgw.net Mon Jun 1 23:05:08 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 16:05:08 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602060508.GA30762@jdc.jasonjgw.net> Also, if the other FS box is behind the same NAT you're on, you should be using the internal profile: sofia/internal/$1 at 192.168.xxx.xxx or whatever. From plite2012 at gmail.com Mon Jun 1 23:12:38 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 2 Jun 2009 01:12:38 -0500 Subject: [Freeswitch-users] How to specify the path to fax file on Windows? Message-ID: All the examples related to faxing use the Unix/Linux path, such as originate sofia/external/100 at 10.10.10.10 &txfax(/path_to_fax_file) I have tried "C:/tmp/fax/txfax.tiff" or "C:\MyJob\fax\txfax.tiff" or "C:\\MyJob\\fax\\txfax.tiff" without any luck. I got an error like [ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file, or the app crashed. Any help is greatly appreciated! From brad.tuan at gmail.com Mon Jun 1 23:18:22 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 14:18:22 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: Like this?? *> *>* *>* * >* *>* * I have tried,but Fs still return: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001 at 192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/50e544d5/attachment.html From brad.tuan at gmail.com Mon Jun 1 23:37:13 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 14:37:13 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: I have tried But FS still return the same message........ [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001 at 192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/231aef43/attachment.html From brad.tuan at gmail.com Tue Jun 2 01:04:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 16:04:49 +0800 Subject: [Freeswitch-users] How to reload xml without using console command line?? Message-ID: As title How to reload xml without using console command line?? -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/db106a0b/attachment.html From jason at jasonjgw.net Tue Jun 2 01:22:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 18:22:56 +1000 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: <20090602082256.GA15770@jdc.jasonjgw.net> Brad Tuan wrote: > As title Write a script that connects to the event socket and issues an api reloadxml command. From jason at jasonjgw.net Tue Jun 2 01:23:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 18:23:56 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602082356.GB15770@jdc.jasonjgw.net> Brad Tuan wrote: > I have tried > > > > Change the % to an @ in the above. From brad.tuan at gmail.com Tue Jun 2 02:00:29 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 17:00:29 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: <20090602082356.GB15770@jdc.jasonjgw.net> References: <20090602082356.GB15770@jdc.jasonjgw.net> Message-ID: the same message........ 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 100 1->97710001 in context default 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot l ocate registered user 97710001 at 192.168.141.187 at internal 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED 2009/6/2 Jason White > Brad Tuan wrote: > > I have tried > > > > > > > > data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/e6737222/attachment.html From krice at suspicious.org Tue Jun 2 02:12:47 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 02 Jun 2009 04:12:47 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: Message-ID: Dumb question... Is 187 the local fs machine? You should have the IP address of the remote FS machine From: Brad Tuan Reply-To: Date: Tue, 2 Jun 2009 17:00:29 +0800 To: Subject: Re: [Freeswitch-users] How to pass a call from one FS to another FS ?? the same message........ ? 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 100 1->97710001 in context default 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot l ocate registered user 97710001 at 192.168.141.187 at internal 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed.? Cause: USER_NOT_REGISTERED ? 2009/6/2 Jason White > Brad Tuan wrote: >> > I have tried >> > >> > >> > ? ? >> > ? ? ?> data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/abfdfc3f/attachment-0001.html From brad.tuan at gmail.com Tue Jun 2 02:25:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 17:25:04 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 These two FS are in the same LAN. I just try to pass one sip call from one FS to another. If it works, next is FS1( PublicIP ) to FS2( PublicIP ). 2009/6/2 Ken Rice > Dumb question... Is 187 the local fs machine? You should have the IP > address of the remote FS machine > > > ------------------------------ > *From: *Brad Tuan > *Reply-To: * > *Date: *Tue, 2 Jun 2009 17:00:29 +0800 > *To: * > *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to > another FS ?? > > > the same message........ > > 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 100 > 1->97710001 in context default > 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() > Cannot l > ocate registered user 97710001 at 192.168.141.187 at internal > 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() > Close Cha > nnel N/A [CS_NEW] > 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Can > not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate > Failed. Cause: USER_NOT_REGISTERED > > 2009/6/2 Jason White > > Brad Tuan wrote: > > I have tried > > > > > > > > data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/3a2eee2a/attachment.html From freeswitch at davidnicol.otherinbox.com Tue Jun 2 03:43:52 2009 From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com) Date: Tue, 2 Jun 2009 06:43:52 -0400 Subject: [Freeswitch-users] Make current fails (build 13537) Message-ID: <200906021043.n52AhqKE005179@box7.911domain.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/af746b3f/attachment.html From shaheryarkh at googlemail.com Tue Jun 2 03:40:02 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 2 Jun 2009 16:40:02 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up Message-ID: Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/ee029b30/attachment.html From yivzhenko at mksat.net Tue Jun 2 04:04:06 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 2 Jun 2009 14:04:06 +0300 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring Message-ID: <200906021404.07496.yivzhenko@mksat.net> Hello all. I have test the lcr overriding the Caller ID functionality. It return dialstring, that contains 'effective_caller_id_number' variable. But that variable has no effect. I try test configuration There is no result. (caller id number not changed) But If I uncomment the set line, then the caller_id_number changes. I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and his status - fixed. ....may be i not consider something? I use svn trunk 13544. Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/780c315c/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 04:08:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 2 Jun 2009 12:08:52 +0100 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: As I understand it, a new 'feature' was added over the weekend to resolve NAT. If you're firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/932019d8/attachment-0001.html From dave at 3c.co.uk Tue Jun 2 04:23:07 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 2 Jun 2009 12:23:07 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: At the risk of evisceration (but with the intention of helping avoid future brain dead build vs. idiot admin debates), I'd suggest that, when significant new bits are added to the switch core, they should default to being off and require a configuration option to turn them on. Such config options can be added to the default config; that way new installs will have the new functionality enabled by default, but those upgrading from an older install will need to enable them manually, reducing the risk of stuff breaking. --Dave ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 01, 2009 11:33 PM Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn?t have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it?s looking for in the logs. That said, to date ?make current? has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it?s very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/c7159f4d/attachment.html From dujinfang at gmail.com Tue Jun 2 04:27:53 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 19:27:53 +0800 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > As I understand it, a new ?feature? was added over the weekend to > resolve NAT. If you?re firewall is not allowing ICMP then FS waits > until it times out. At this time there is no option to disable it. > > Regards > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Muhammad Shahzad > Sent: 02 June 2009 11:40 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Freeswitch taking too long to start up > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I > am using 32bit CentOS 5.3, "make current" command completes > successfully without any errors but when i start freeswitch it take > considerable time (roughly 90 - 120 seconds) to start up. During > this time no message is display on console. Once successfully > started, it works fine. However, this initial delay is really > annoying. Is there anyway to reduce/remove this delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/468e97b7/attachment.html From brad.tuan at gmail.com Tue Jun 2 04:28:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 19:28:04 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call Message-ID: Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"?? EVENT DUMP: Channel-State: [CS_ROUTING] Channel-State-Number: [2] Channel-Name: [sofia/internal/sip:97730001 at 210.68.184.192:62101 ;rinstance=16b8076934af7da9] Unique-ID: [342618e3-84cd-494b-b745-760b60639924] Call-Direction: [outbound] Answer-State: [ringing] Caller-Username: [97719006] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Extension 97730002] Caller-Caller-ID-Number: [97730002] Caller-Network-Addr: [163.28.32.51] Caller-Destination-Number: [sip:97730001 at 210.68.184.192:62101 ;rinstance=16b80769 34af7da9] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/879ab698/attachment.html From codecomplete at free.fr Tue Jun 2 04:40:55 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 2 Jun 2009 04:40:55 -0700 (PDT) Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <4A23D15E.3040908@coppice.org> References: <23807353.post@talk.nabble.com> <4A23D15E.3040908@coppice.org> Message-ID: <23830858.post@talk.nabble.com> Steve Underwood wrote: > Nobody has yet adapted Freeswitch for the Blackfin, and they probably > won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux, > which is a cut down Linux for machines of this type. It is quite > troublesome to get memory management to behave sanely on a machine > without an MMU. The Asterisk adaptions for the Blackfin have problems > with this too, but if you don't let the memory become too fragmented they > work OK. Thanks much for the explanation. I don't need the fax module. Hopefully, other features will work fine on this unit. I haven't found other hardware that is as compact and affordable as the Atcom while providing an embedded FXO port. -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p23830858.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From juanbackson at gmail.com Tue Jun 2 05:01:05 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 2 Jun 2009 20:01:05 +0800 Subject: [Freeswitch-users] some fifo questions Message-ID: <27c25bc40906020501xd51fea2x49949b8855306751@mail.gmail.com> Hi, I read the fifo section of the wiki and what is not clear are: What is the meaning of fifo_orbit_announce? What is the meaning of fifo_override_announce? Is it possible to create a scenario where the caller can hear "Agent #123 is going to attend to your call"? Any help will be greatly appreciated. Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/02c2ec6c/attachment-0001.html From jim at evolutiontel.net Tue Jun 2 05:25:38 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 2 Jun 2009 22:25:38 +1000 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> <191c3a030905280607q7dd0b5b7xc9e3f02b9f6c8824@mail.gmail.com> <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> Message-ID: Hey Gents, What is the Jira for this issue? Dale, Did you get any SIP traces. I am interested to have a look. You can use NGREP if your system is Linux. Regards, Jim On Thu, May 28, 2009 at 11:08 PM, Anthony Minessale wrote: > btw, > > ?3 and 4 are not useful without 1 > we only debug issues with svn trunk > > > On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale > wrote: >> >> Also you should be putting these details in a jira report. >> http://jira.freeswitch.org >> >> open an issue report and attach all relevant logs, do not attach tarballs >> or gzipped files and make sure text files have a .txt extension. >> >> >> On Wed, May 27, 2009 at 6:58 PM, Dale Trub wrote: >>> >>> Anthony, >>> >>> Thank you for your suggestions!? We are working on 1), but need to >>> re-integrate code we've changed, and do regression testing. That's in >>> progress, and we expect to be able to upgrade by the end of next week. >>> >>> We did manage to do 3) and 4), and we now have SIP logs (attached). Are >>> you able to see anything that's out of the ordinary that we should be paying >>> attention to? >>> >>> Best, >>> Dale >>> >>> On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale >>> wrote: >>>> >>>> 1) update to lastest trunk (you are at least 1000 revisions behind) >>>> 2) disable the presence debug in sofia.conf >>>> 3) enable sip trace instead "sofia profile internal siptrace on" >>>> 4) reproduce your problem. >>>> >>>> Make sure you include more of the log from before the hangup happened. >>>> The one you posted here is missing some of the info from the few seconds >>>> prior but with the incomplete >>>> info it looks like the other side sent a BYE ending the call. >>>> >>>> >>>> On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: >>>>> >>>>> Thanks Brian! ?To answer your questions: >>>>> Freeswitch svn revision: 12148 >>>>> Centos rev: 2.6.18-92.el5 >>>>> And apologies, actually I guess we're using g711 not 729. >>>>> Jason: ?I agree it would seem to be on the switch/telco side. ?And, the >>>>> telco says many other people are in the same set-up as us and don't have any >>>>> issues, so they're insisting it's on our end. >>>>> On Thu, May 21, 2009 at 7:28 PM, Brian West >>>>> wrote: >>>>>> >>>>>> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >>>>>> >>>>>> We're running FreeSwitch as part of a teleconferencing service, inside >>>>>> a telcom?(so no >>>>>> internet latency/NAT issues)?and using g.729 >>>>>> >>>>>> So you're using g729 with conferences? >>>>>> >>>>>> We are?receiving some complaints of dropped calls, >>>>>> including from landlines. ? This means they join the conference, and x >>>>>> minutes in they simply drop. >>>>>> I?know that cellphones tend to drop calls frequently, but landlines >>>>>> are pretty reliable, and we're hearing it a lot. ?From the FreeSwitch >>>>>> side of things, it just >>>>>> looks like those callers hung up (but then dialed back in just a >>>>>> moment later). >>>>>> I'm attaching two different snippets of the FS log files where these >>>>>> issues are occurring. >>>>>> >>>>>> Next time please call them .txt because you cause extra work to have >>>>>> to open them otherwise. >>>>>> >>>>>> Does anyone have any recommendations about how to troubleshoot this? >>>>>> Any known issues/patches in FS that could be biting us? >>>>>> >>>>>> Depends you failed to include some very valid info such as what >>>>>> version or svn rev you're running and what linux distro. >>>>>> >>>>>> Is there some SIP logging we can do to debug? >>>>>> >>>>>> Yes covered on the wiki. >>>>>> ?http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>>>> >>>>>> Are there any paid contractors avail who would have the expertise to >>>>>> look into this? >>>>>> >>>>>> email consulting at freeswitch.org >>>>>> >>>>>> Any help appreciated ... this is a major issue for us! >>>>>> Thanks much, >>>>>> -Dale >>>>>> >>>>>> Brian West >>>>>> brian at freeswitch.org >>>>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Tue Jun 2 05:38:30 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 2 Jun 2009 18:38:30 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: > Actually Brain mentioned that you can comment out switch_nat_init(); in > switch_core.c > On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > > As I understand it, a new ?feature? was added over the weekend to resolve > NAT. If you?re firewall is not allowing ICMP then FS waits until it times > out. At this time there is no option to disable it. > > Regards > > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Muhammad Shahzad > *Sent:* 02 June 2009 11:40 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch taking too long to start up > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I am > using 32bit CentOS 5.3, "make current" command completes successfully > without any errors but when i start freeswitch it take considerable time > (roughly 90 - 120 seconds) to start up. During this time no message is > display on console. Once successfully started, it works fine. However, this > initial delay is really annoying. Is there anyway to reduce/remove this > delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/3fccb446/attachment.html From brian at freeswitch.org Tue Jun 2 05:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 07:52:10 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: Its coming soon! /b On Jun 2, 2009, at 6:23 AM, David Knell wrote: > At the risk of evisceration (but with the intention of helping avoid > future brain dead build vs. idiot admin debates), I'd suggest that, > when significant new bits are added to the switch core, they should > default to being off and require a configuration option to turn them > on. Such config options can be added to the default config; that > way new installs will have the new functionality enabled by default, > but those upgrading from an older install will need to enable them > manually, reducing the risk of stuff breaking. > > --Dave Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/fe248b97/attachment.html From brian at freeswitch.org Tue Jun 2 05:52:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 07:52:46 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: Chances are that is what you set it to on the user. Verify the users settings in the directory. /b On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: > Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name > is "Extension 97730002"?? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/df2565bb/attachment-0001.html From brad.tuan at gmail.com Tue Jun 2 06:11:07 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:11:07 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: I don't have a 97719006 User in my FS. It was passed from another sip proxy. 2009/6/2 Brian West > Chances are that is what you set it to on the user. Verify the users > settings in the directory. > /b > > On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: > > Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is > "Extension 97730002"?? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/b8f033b0/attachment.html From brian at freeswitch.org Tue Jun 2 06:26:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 08:26:16 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: Then it had to be passed in from the proxy. /b On Jun 2, 2009, at 8:11 AM, Brad Tuan wrote: > I don't have a 97719006 User in my FS. > > It was passed from another sip proxy. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/70456bce/attachment.html From brad.tuan at gmail.com Tue Jun 2 06:28:58 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:28:58 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: When send 100 Trying: From: 97719006 ;tag=124388224932run00 But when send INVITE: From: "Extension 97730002" >;tag=U3QF8QUp1F3tQ What happened between sending Trying and sending INVITE ?? ------------------------------------------------------------------------ send 556 bytes to udp/[163.28.32.51]:5070 at 13:03:05.218750: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK5b64.83b3e3d3.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12439477847278735900 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124394778472787359 Record-Route: From: 97719006 ;tag=124388224932run00 To: Call-ID: i58YWNjMDU3ZWJhN2M1YzVlYjMzOTgxMjk4OWZiNTU0Yzc.00 CSeq: 7359 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/97719006 at 61.61.162.13 -b28d-fd1ecda44f18] 2009-06-02 21:03:05 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 97719006->97730001 in context default 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::C:SipG -06-02-21-03-05.wav 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:97730001 at 210.68.1 d6224d69efcf1 [e8ef86f8-e5aa-d246-8ffb-bf9e0f9dc160] send 1347 bytes to udp/[210.68.184.192]:62113 at 13:03:05.812500: ------------------------------------------------------------------------ INVITE sip:97730001 at 210.68.184.192:62113;rinstance=d0ed6224d69efcf1 SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKg01ymegmSyp5c Max-Forwards: 67 From: "Extension 97730002" >;tag=U3QF8QUp1F3tQ To: Call-ID: 8dcc44c8-ca18-122c-2780-39a48cb53b8d CSeq: 115855556 INVITE 2009/6/2 Brad Tuan > I don't have a 97719006 User in my FS. > > It was passed from another sip proxy. > > > 2009/6/2 Brian West > >> Chances are that is what you set it to on the user. Verify the users >> settings in the directory. >> /b >> >> On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: >> >> Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is >> "Extension 97730002"?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/10c48ae7/attachment.html From brian at freeswitch.org Tue Jun 2 06:33:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 08:33:05 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: I would update if I were you! :) Anyway something had to have changed it it won't magically do it. /b On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/31589174/attachment.html From brad.tuan at gmail.com Tue Jun 2 06:38:45 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:38:45 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: So I need to new a User(97719006) in directory\default ?? 2009/6/2 Brian West > I would update if I were you! :) Anyway something had to have changed it > it won't magically do it. > /b > > On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/fc31d939/attachment-0001.html From brad.tuan at gmail.com Tue Jun 2 06:41:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:41:04 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: How to update FreeSWITCH-mod_sofia/1.0.3-12163?? 2009/6/2 Brian West > I would update if I were you! :) Anyway something had to have changed it > it won't magically do it. > /b > > On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f4bcc499/attachment.html From rex.alex345 at yahoo.com Tue Jun 2 07:14:13 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 2 Jun 2009 07:14:13 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS Message-ID: <1243952053200-3012286.post@n2.nabble.com> Hello, I have installed FS, and tested outbound successfully. Now I am just trying to do the inbound testing. I got the Inbound DID. Please suggest me what changes should I make and where? Thanks, Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012286.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/bdc5fce0/attachment.html From dujinfang at gmail.com Tue Jun 2 07:27:00 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:27:00 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: Hi, I always got 0 messages when using web. Finally I added some debug information in the code and get this: 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3883 voicemail_api_function() port:[8080] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3884 voicemail_api_function() uri:[/domains/192.168.1.16/api/voicemail] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3885 voicemail_api_function() user:[] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3886 voicemail_api_function() domain:[192.168.1.16] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3887 voicemail_api_function() path_info:[web] It seems the freeswitch-user header not set by xml_rpc and user = switch_event_get_header(stream->param_event, "freeswitch-user"); cannot get the user. Any idea? Thanks. From dujinfang at gmail.com Tue Jun 2 07:30:27 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:30:27 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: <35216FF7-6120-4F44-84B5-25F1540A93C3@gmail.com> sorry forgot to mention I'm on FreeSWITCH Version 1.0.trunk (13524M) From d at unwire.it Tue Jun 2 07:35:59 2009 From: d at unwire.it (Darin Weeks) Date: Tue, 2 Jun 2009 07:35:59 -0700 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243952053200-3012286.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> Message-ID: <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> Have you setup an inbound gateway similar to this? http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing You also need to setup your dialplans for the inbound.... this page among others has more info: http://wiki.freeswitch.org/wiki/Quick_Start Finally, your FIREWALL can be the most critical to get right. In fact, you might want to start with this or revisit even if you think it is setup right. I was grasping around in the dark until I started using TCPDUMP to monitor what was happening with my connections. In the end, I realized that I needed to open certain ports on the SENDING side -- so, for example, calls coming FROM port 5060 to *any* port on my side I actually forward to port 5080 on my freeswitch server. At least I think that's what I ended up doing, but there are several different rules I setup as well. On Tue, Jun 2, 2009 at 7:14 AM, Rex_Alex wrote: > Hello, I have installed FS, and tested outbound successfully. Now I am just > trying to do the inbound testing. I got the Inbound DID. Please suggest me > what changes should I make and where? Thanks, Rex > ------------------------------ > View this message in context: Inbound using FS > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/4dd04e3a/attachment.html From dujinfang at gmail.com Tue Jun 2 07:36:34 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:36:34 +0800 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243952053200-3012286.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> Message-ID: <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> I would route the DID to the host and port 5080 if you are using the default config, and make an extension in dialplan/public.xml to catch the DID. Press F8 to see the debug information if not sure what DID string should be matched. On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote: > Hello, I have installed FS, and tested outbound successfully. Now I > am just trying to do the inbound testing. I got the Inbound DID. > Please suggest me what changes should I make and where? Thanks, Rex > View this message in context: Inbound using FS > Sent from the freeswitch-users mailing list archive at Nabble.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/dbddbbe7/attachment.html From d at unwire.it Tue Jun 2 07:39:00 2009 From: d at unwire.it (Darin Weeks) Date: Tue, 2 Jun 2009 07:39:00 -0700 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> Message-ID: <989132e70906020739n2e7084dm2f3e57d1555f20e2@mail.gmail.com> UPDATE: I just looked at my firewall rules and looks like I scrapped all the logic I was attempting and now I just port forward ANYTHING coming from the IP of my provider gateway to my freeswitch box. Seems to be working fine. On Tue, Jun 2, 2009 at 7:35 AM, Darin Weeks wrote: > Have you setup an inbound gateway similar to this? > http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing > > You also need to setup your dialplans for the inbound.... this page among > others has more info: > http://wiki.freeswitch.org/wiki/Quick_Start > > Finally, your FIREWALL can be the most critical to get right. In fact, you > might want to start with this or revisit even if you think it is setup > right. I was grasping around in the dark until I started using TCPDUMP to > monitor what was happening with my connections. In the end, I realized that > I needed to open certain ports on the SENDING side -- so, for example, calls > coming FROM port 5060 to *any* port on my side I actually forward to port > 5080 on my freeswitch server. At least I think that's what I ended up > doing, but there are several different rules I setup as well. > > On Tue, Jun 2, 2009 at 7:14 AM, Rex_Alex wrote: > >> Hello, I have installed FS, and tested outbound successfully. Now I am >> just trying to do the inbound testing. I got the Inbound DID. Please suggest >> me what changes should I make and where? Thanks, Rex >> ------------------------------ >> View this message in context: Inbound using FS >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/6aa5e361/attachment-0001.html From mike at jerris.com Tue Jun 2 08:59:32 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Jun 2009 11:59:32 -0400 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: /usr/local/freeswitch/bin/fs_cli -x reloadxml On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote: > How to reload xml without using console command line?? From mike at jerris.com Tue Jun 2 09:04:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Jun 2009 12:04:53 -0400 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <200906021404.07496.yivzhenko@mksat.net> References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: <07772B7F-6DCD-4D70-A3F9-AE861CEB6E29@jerris.com> can you try in the square brackets using http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number instead? I think effective will work if you set it but not in the square brackets. Mike On Jun 2, 2009, at 7:04 AM, Yuriy Ivzhenko wrote: > Hello all. > I have test the lcr overriding the Caller ID functionality. > It return dialstring, that contains 'effective_caller_id_number' > variable. > But that variable has no effect. > I try test configuration > > > > data="[effective_caller_id_number=9999]sofia/internal/sip:1001 at 192.168.2.43:5060 > "/> > > > > There is no result. (caller id number not changed) > But If I uncomment the set line, then the caller_id_number changes. > I found the similar bug (http://jira.freeswitch.org/browse/ > MODAPP-122) and his status - fixed. > ....may be i not consider something? > I use svn trunk 13544. > Yuriy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/7c3ddb15/attachment.html From rex.alex345 at yahoo.com Tue Jun 2 09:11:50 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 2 Jun 2009 09:11:50 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> Message-ID: <1243959110713-3012928.post@n2.nabble.com> Hello, My public.xml configration is: My default.xml configration is: When I am trying to call 123456 from my mobile no. Not able to see any logging in FS console. Please assist where I am going wrong? Or do I require any extra modules to be installed? Thanks, Rex dujinfang wrote: > > I would route the DID to the host and port 5080 if you are using the > default config, and make an extension in dialplan/public.xml to catch > the DID. Press F8 to see the debug information if not sure what DID > string should be matched. > > > On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote: > >> Hello, I have installed FS, and tested outbound successfully. Now I >> am just trying to do the inbound testing. I got the Inbound DID. >> Please suggest me what changes should I make and where? Thanks, Rex >> View this message in context: Inbound using FS >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012928.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 2 09:15:52 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 11:15:52 -0500 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243959110713-3012928.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> Message-ID: On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > > Hello, > > My public.xml configration is: > > > > > > $1 will not exist in this case because your regular expression doesn't capture anything. So replace $1 with your target number or use ^(123456)$ > > My default.xml configration is: > > > > > > > > Can you elaborate how you're registering with your provider? > > > When I am trying to call 123456 from my mobile no. Not able to see any > logging in FS console. Please assist where I am going wrong? Or do I > require > any extra modules to be installed? > > Thanks, > Rex Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/98113acd/attachment.html From rdenert at tng.de Tue Jun 2 09:18:49 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 2 Jun 2009 18:18:49 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <2613803.271261243959087777.JavaMail.root@zimbra.tng.de> Message-ID: <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Greetz From dujinfang at gmail.com Tue Jun 2 09:30:14 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 00:30:14 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: I thought it is a problem, made a jira: http://jira.freeswitch.org/browse/XML-2 From rupa at rupa.com Tue Jun 2 09:40:12 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 2 Jun 2009 11:40:12 -0500 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <200906021404.07496.yivzhenko@mksat.net> References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: I've fixed mod_lcr now. It should have been setting origination_caller_id_number not effective_caller_id_number. On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > Hello all. > > I have test the lcr overriding the Caller ID functionality. > > It return dialstring, that contains 'effective_caller_id_number' variable. > > But that variable has no effect. > > I try test configuration > > > > > > > > data="[effective_caller_id_number=9999]sofia/internal/ > sip:1001 at 192.168.2.43:5060"/> > > > > > > > > There is no result. (caller id number not changed) > > But If I uncomment the set line, then the caller_id_number changes. > > I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and > his status - fixed. > > ....may be i not consider something? > > I use svn trunk 13544. > > Yuriy > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/14779423/attachment.html From anthony.minessale at gmail.com Tue Jun 2 09:50:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 11:50:13 -0500 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> effective_* is *NOT EVER* valid in the dial string. they are settings of an existing session to control what caller id they pass. On Tue, Jun 2, 2009 at 11:40 AM, Rupa Schomaker wrote: > I've fixed mod_lcr now. It should have been setting > origination_caller_id_number not effective_caller_id_number. > > On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > >> Hello all. >> >> I have test the lcr overriding the Caller ID functionality. >> >> It return dialstring, that contains 'effective_caller_id_number' variable. >> >> But that variable has no effect. >> >> I try test configuration >> >> >> >> >> >> >> >> > data="[effective_caller_id_number=9999]sofia/internal/ >> sip:1001 at 192.168.2.43:5060"/> >> >> >> >> >> >> >> >> There is no result. (caller id number not changed) >> >> But If I uncomment the set line, then the caller_id_number changes. >> >> I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) >> and his status - fixed. >> >> ....may be i not consider something? >> >> I use svn trunk 13544. >> >> Yuriy >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f69dca62/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 10:04:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 2 Jun 2009 18:04:14 +0100 Subject: [Freeswitch-users] Outbound socket question Message-ID: Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f68fdd95/attachment.html From anthony.minessale at gmail.com Tue Jun 2 10:18:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 12:18:29 -0500 Subject: [Freeswitch-users] Outbound socket question In-Reply-To: References: Message-ID: <191c3a030906021018t5f913b49u593c286bc7324d64@mail.gmail.com> yes the socket remains open the duration of your connection. and the uuid becomes optional at that point for sendmsg but may still come into play for some FSAPI based commands like uuid_getvar On Tue, Jun 2, 2009 at 12:04 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?m going some work with outbound socket, and have a few questions. > > > > When each call is answered, I get a connection to my server socket. > > > > Is it right to assume that this connection will remain for the duration of > the call? > > > > If so, do I still need to pass the UUID when I call an application such as > playfile? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/b9436296/attachment.html From dave at 3c.co.uk Tue Jun 2 10:26:30 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 2 Jun 2009 18:26:30 +0100 Subject: [Freeswitch-users] Outbound socket question References: Message-ID: <2206B39F96274B17B468DF2634FBB012@DELL9> Hi Nik, Yes and no, respectively. Cheers -- Dave ----- Original Message ----- From: Nik Middleton To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 02, 2009 6:04 PM Subject: [Freeswitch-users] Outbound socket question Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/095e94bb/attachment-0001.html From msc at freeswitch.org Tue Jun 2 11:02:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:02:53 -0700 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: > How to update FreeSWITCH-mod_sofia/1.0.3-12163?? Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible. Let us know how it goes... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/de577ef7/attachment.html From msc at freeswitch.org Tue Jun 2 11:18:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:18:41 -0700 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> References: <2613803.271261243959087777.JavaMail.root@zimbra.tng.de> <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906021118g18fdd313n66625e5f414d4fe8@mail.gmail.com> On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert wrote: > Hello, > > I have problems with sending DTMF. The freeswitch server recieves the > digits from my telefon (SIP account) if I activate DTMF with: > > in my dialplan. > > The line > <--param name="dtmf-type" value="rfc2833"/--> > in sofia.conf.xml is active, too. > > But than I have the problem that the other phone doesn't work. It is a > VoATM device. The curious thing is that I see the digits in the logfile > whiche were sent from the phone. In the first example I saw nothing. > > Does anybody have an idea??? > Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/dce80840/attachment.html From msc at freeswitch.org Tue Jun 2 11:14:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:14:49 -0700 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> References: <200906021404.07496.yivzhenko@mksat.net> <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> Message-ID: <87f2f3b90906021114i70b12db5h62a7d021b9091eb2@mail.gmail.com> On Tue, Jun 2, 2009 at 9:50 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > effective_* is *NOT EVER* valid in the dial string. they are settings of > an existing session to control what caller id they pass. > > FYI, I've updated the wiki to reflect this fact and to make it completely obvious: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/887f0f10/attachment.html From rdenert at tng.de Tue Jun 2 11:37:14 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 2 Jun 2009 20:37:14 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <14151206.271881243967650954.JavaMail.root@zimbra.tng.de> Message-ID: <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From keithl at voxtelecom.co.za Tue Jun 2 12:12:20 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Tue, 2 Jun 2009 21:12:20 +0200 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Hi, Try starting using the -nonat switch. Best Regards Keith From: Muhammad Shahzad [mailto:shaheryarkh at googlemail.com] Sent: 02 June 2009 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch taking too long to start up Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: As I understand it, a new ?feature? was added over the weekend to resolve NAT. If you?re firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/62270a27/attachment-0001.html From msc at freeswitch.org Tue Jun 2 12:29:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 12:29:19 -0700 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> References: <14151206.271881243967650954.JavaMail.root@zimbra.tng.de> <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906021229v5784466dv2932aaa162c052d8@mail.gmail.com> On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert wrote: > Hello, > > I'm not sure which one is it. But I think I send the digits in RFC 2833. > All devicves are supporting RFC 2833. > Actually, that's very interesting. The start_dtmf app looks for in-band DTMFs, so if you have to have that in order to hear digits from the other end then the other end most definitely isn't doing RFC2833. For the sake of testing, try sending in-band and see how the other end reacts. Might want to check this out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate My guess is that the equipment along the way is futzing with things. FreeSWITCH does break easily but it does find bugs in other VoIP systems that it talks to... :) -MC > > The equipment: > (VoATM) > Allied Data Copperjet 1614 (ISDN) > Siemens Euroset 5020 phone > > (MGCP) > Thomson SpeedTouch 780WL > Siemens Euroset 5020 phone > > (SIP) > AVM Fritz!Box 7170 > Siemens Euroset 5020 phone > > If necessary I can send my configuration. > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Michael Collins" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 > Amsterdam/Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Prolems with DTMF > > > > > > On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: > > > Hello, > > I have problems with sending DTMF. The freeswitch server recieves the > digits from my telefon (SIP account) if I activate DTMF with: > > in my dialplan. > > The line > <--param name="dtmf-type" value="rfc2833"/--> > in sofia.conf.xml is active, too. > > But than I have the problem that the other phone doesn't work. It is a > VoATM device. The curious thing is that I see the digits in the logfile > whiche were sent from the phone. In the first example I saw nothing. > > Does anybody have an idea??? > > > Are you trying to send digits inband or RFC2833? Unless there's a > compelling reason not to, we recommend 2833. What is the equipment on the > far end looking for? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/71075867/attachment.html From dome at tel.co.th Tue Jun 2 12:42:27 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 3 Jun 2009 02:42:27 +0700 Subject: [Freeswitch-users] How to change sound-path when switch language Message-ID: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> Dear sir, i create mod_say_th for Thai language. i found some problem about sound-path. I have config th.xml in conf/lang/th/ ... when i try Freeswitch still looking sounf file in /sounds/en/us/callie (en sound-path) Someone help me please Dome C. From brian at freeswitch.org Tue Jun 2 12:50:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 14:50:06 -0500 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> Message-ID: <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> You'll need to set the variable default_language /b On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote: > Dear sir, > i create mod_say_th for Thai language. i found some problem > about sound-path. > I have config th.xml in conf/lang/th/ > tts-engine="cepstral" tts-voice="callie"> > ... > > when i try > > Freeswitch still looking sounf file in /sounds/en/us/callie (en > sound-path) > > Someone help me please Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/87eb8083/attachment.html From fvillarroel at yahoo.com Tue Jun 2 13:46:46 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Tue, 2 Jun 2009 13:46:46 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <302665.69883.qm@web34301.mail.mud.yahoo.com> Dear, I can't solve my problem, i was try with: and: in freeswitch.xml But receive the same log: http://pastebin.freeswitch.org/9204 Anyone help me. Fernando --- On Mon, 6/1/09, FERNANDO VILLARROEL wrote: > From: FERNANDO VILLARROEL > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 8:10 PM > > Hello i was try with: > > > data="sofia/gateway/ubb/$1$2$3"/> > > This is the log on FS_CLI: > > http://pastebin.freeswitch.org/9204 > > Fernando > > --- On Mon, 6/1/09, Michael Collins > wrote: > > > From: Michael Collins > > Subject: Re: [Freeswitch-users] Passthru mode > > To: freeswitch-users at lists.freeswitch.org > > Date: Monday, June 1, 2009, 7:41 PM > > > > > > On Mon, Jun 1, 2009 at 3:20 PM, > > FERNANDO VILLARROEL > > wrote: > > > > > > > > Hello the dial plan: > > > > > > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > > > > > This i setup from Wikipbx. > > What about this in the dialplan? > > > data="proxy_media=true"/> > > Or alternatively this in the SIP profile? > > > > > value="true"/> > > > > I just want to make sure you're actually telling FS > to > > use proxy media. If I may make a suggestion: use > pastebin.freeswitch.org > > and pastebin the entire extension in the dialplan as > well as > > a complete debug log of the call from the FS CLI. > Please see > > this page for some handy tips on gathering information > for > > troubleshooting: > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > -MC > > > > > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ? ? ? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From larclap at yahoo.com Tue Jun 2 14:53:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 2 Jun 2009 14:53:38 -0700 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: <035b01c9e3cc$96486450$c2d92cf0$@com> Brian, I'm probably not the only one here, but much of what I have to do to get Freeswitch going is new to me. Never installed or really worked with Linux and scripting; just a little xml. It is challenging. Freeswitch is interesting, appealing and challenging. The work your group has done is amazing. Given this, interacting with you can be intimidating. I am experiencing the slow start with build 13532. I assume that "block all ICMP" refers to the firewall/gateway. If this is correct, why is it that I can ping the firewall from the Freeswitch box? Can you explain in more detail what it might be on my network that is blocking ICMP? All my clients and Freeswitch itself are behind a NAT firewall. Thanks Lars Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 01, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/bc729270/attachment-0001.html From brian at freeswitch.org Tue Jun 2 15:01:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 17:01:51 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <035b01c9e3cc$96486450$c2d92cf0$@com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: <315C26CB-8738-46E4-A2CD-AB812BBB9200@freeswitch.org> We are working to correct it. So hold on ;) /b On Jun 2, 2009, at 4:53 PM, Lars Zeb wrote: > Brian, > > I?m probably not the only one here, but much of what I have to do to > get Freeswitch going is new to me. Never installed or really worked > with Linux and scripting; just a little xml. It is challenging. > Freeswitch is interesting, appealing and challenging. The work your > group has done is amazing. Given this, interacting with you can be > intimidating. > > I am experiencing the slow start with build 13532. I assume that > ?block all ICMP? refers to the firewall/gateway. If this is correct, > why is it that I can ping the firewall from the Freeswitch box? Can > you explain in more detail what it might be on my network that is > blocking ICMP? All my clients and Freeswitch itself are behind a NAT > firewall. > > Thanks Lars > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 > i686 i386 GNU/Linux Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f2b02bb4/attachment.html From anthony.minessale at gmail.com Tue Jun 2 15:14:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 17:14:29 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> You have a good point. On the other hand, it's just another random day in SVN trunk. =D Most projects don't offer SVN trunk you can play spin-the-bottle with and land on something production-ready. But we are pretty close most of the time. Here's my point of view: That particular addition was a component to the core meant to be transparent. If we did not find out the hard-way about this by adding it to trunk, we would have found out the even-harder-way by having it imprinted in the actual release. We try to keep the suffering to a minimum but we sometimes fall short. On Tue, Jun 2, 2009 at 6:23 AM, David Knell wrote: > At the risk of evisceration (but with the intention of helping avoid > future brain dead build vs. idiot admin debates), I'd suggest that, when > significant new bits are added to the switch core, they should default to > being off and require a configuration option to turn them on. Such config > options can be added to the default config; that way new installs will have > the new functionality enabled by default, but those upgrading from an older > install will need to enable them manually, reducing the risk of stuff > breaking. > > --Dave > > ----- Original Message ----- > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, June 01, 2009 11:33 PM > *Subject:* Re: [Freeswitch-users] Make current fails (build 13537) > > NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since > your network must be eating the packets its sending out to detect if you're > behind nat or not... and not getting an ICMP unreachable like it should be > getting... the joys of admins that block all ICMP like idiots. ICMP has > many uses... and outright blocking it is stupid. (This is my assumption cuz > its what makes sense in this case) > So you're getting hit by the nice retry/timeout loop in the natpmp software > we just added and possibly the upnp lib too. > > So for now edit switch_core.c and comment out switch_nat_init(); > > I'm working my ass off to ensure that our users that do have to live in > these insane nat scenarios can do so without much if any pain. Most of which > uses SMB/Consumer grade routers which these two libs we added would allow us > to poke holes and setup stuff and make it painless as possible. > > Soon you'll have an option in switch.conf.xml to turn it off. > > Please next time don't be so demanding and calling builds brain dead .. > when in fact its trying to become more aware of its network config without > much user input. > > /b > > On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: > > Well I can only assume build 13537 is brain dead. Surely I shouldn?t > have to edit a whole bunch of configs to get it working. FS now takes 3 > minutes to start, with no indication as to what it?s looking for in the > logs. That said, to date ?make current? has always worked well for me. > Guess I was bound to hit a bad one eventually. > Still, it?s very frustrating. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/8f036afb/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 16:37:27 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 3 Jun 2009 00:37:27 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <035b01c9e3cc$96486450$c2d92cf0$@com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: As Anthony comments later, using SVN for updates is usually a risky business for most projects. We all have been blessed by fantastic coding to date with this project, that has lulled us into believing that using the latest snapshot will be OK. This is the first time that I've had problems. I have no doubt that the DEV's have taken this onboard, but it can sometimes be a reality check to realize that the subscribed based has grown to such a size that regression testing now becomes mandatory if the project is to move onto the next stage. A very valid comment was made on this thread that new features should be disabled by default until thoroughly tested. It's all part of the learning cycle. In my view the trunk needs to be updated more frequently and this should be what us mere mortals use. To often I see messages saying you're using a 2 week old version, that bug's been fixed. FS, is coming to a level where code has to be managed in a more structured way, but I have now doubt this will be addressed fairly rapidly. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: 02 June 2009 22:54 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) Brian, I'm probably not the only one here, but much of what I have to do to get Freeswitch going is new to me. Never installed or really worked with Linux and scripting; just a little xml. It is challenging. Freeswitch is interesting, appealing and challenging. The work your group has done is amazing. Given this, interacting with you can be intimidating. I am experiencing the slow start with build 13532. I assume that "block all ICMP" refers to the firewall/gateway. If this is correct, why is it that I can ping the firewall from the Freeswitch box? Can you explain in more detail what it might be on my network that is blocking ICMP? All my clients and Freeswitch itself are behind a NAT firewall. Thanks Lars Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 01, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/d651d2f1/attachment-0001.html From msc at freeswitch.org Tue Jun 2 16:38:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 16:38:21 -0700 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> Message-ID: <87f2f3b90906021638t51d3dfcdo40426d2f13c22ffc@mail.gmail.com> On Tue, Jun 2, 2009 at 3:14 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You have a good point. > > On the other hand, it's just another random day in SVN trunk. =D > Most projects don't offer SVN trunk you can play spin-the-bottle with and > land on something production-ready. But we are pretty close most of the > time. > > Here's my point of view: > That particular addition was a component to the core meant to be > transparent. > If we did not find out the hard-way about this by adding it to trunk, > we would have found out the even-harder-way by having it imprinted in the > actual release. > > We try to keep the suffering to a minimum but we sometimes fall short. > This is also why we need as many people as possible updating FS as often as possible. The greater the number of environments we have running FreeSWITCH, the less likely it is that stuff like this will sneak through and the more likely it will be caught and fixed quickly. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/a273254f/attachment.html From brian at freeswitch.org Tue Jun 2 16:46:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 18:46:10 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: You now have -nonat and the hang on start up with the nat detection code is fixed now. /b On Jun 2, 2009, at 6:37 PM, Nik Middleton wrote: > As Anthony comments later, using SVN for updates is usually a risky > business for most projects. We all have been blessed by fantastic > coding to date with this project, that has lulled us into believing > that using the latest snapshot will be OK. This is the first time > that I?ve had problems. > > I have no doubt that the DEV?s have taken this onboard, but it can > sometimes be a reality check to realize that the subscribed based > has grown to such a size that regression testing now becomes > mandatory if the project is to move onto the next stage. > > A very valid comment was made on this thread that new features > should be disabled by default until thoroughly tested. It?s all part > of the learning cycle. In my view the trunk needs to be updated > more frequently and this should be what us mere mortals use. To > often I see messages saying you?re using a 2 week old version, that > bug?s been fixed. > > FS, is coming to a level where code has to be managed in a more > structured way, but I have now doubt this will be addressed fairly > rapidly. > > Regards, > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/40b69769/attachment.html From brad.tuan at gmail.com Tue Jun 2 17:41:52 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 08:41:52 +0800 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: Thanks a lot ! This's what i want. 2009/6/2 Michael Jerris > /usr/local/freeswitch/bin/fs_cli -x reloadxml > > On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote: > > > How to reload xml without using console command line?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/e2f62f36/attachment.html From dujinfang at gmail.com Tue Jun 2 19:20:43 2009 From: dujinfang at gmail.com (seven) Date: Wed, 3 Jun 2009 10:20:43 +0800 Subject: [Freeswitch-users] Is there a way to cancel att_xfer? Message-ID: <0FC73ACB-C36D-4CE3-A2C6-7E3CB9AD63C8@gmail.com> Hi, Assume the following sinario: A call B, B att_xfer to C if no answer on C for a long time, B can cancel the att_xfer by pressing a key and talk to A again. Is that possible? Thank you. 7. From brad.tuan at gmail.com Tue Jun 2 19:49:15 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 10:49:15 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: I was updated my FS and rebuilt it. It works........ But when User2(FS2) accept the call from User1(FS1) , User2(FS2) display "call established",but User1(FS1) still display "calling". Why?? (I think maybe that I need to do some setting on FS2.) 2009/6/2 Brad Tuan > FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 > > These two FS are in the same LAN. > > I just try to pass one sip call from one FS to another. > > If it works, next is FS1( PublicIP ) to FS2( PublicIP ). > > 2009/6/2 Ken Rice > > Dumb question... Is 187 the local fs machine? You should have the IP >> address of the remote FS machine >> >> >> ------------------------------ >> *From: *Brad Tuan >> *Reply-To: * >> *Date: *Tue, 2 Jun 2009 17:00:29 +0800 >> *To: * >> *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to >> another FS ?? >> >> >> the same message........ >> >> 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 100 >> 1->97710001 in context default >> 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() >> Cannot l >> ocate registered user 97710001 at 192.168.141.187 at internal >> 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() >> Close Cha >> nnel N/A [CS_NEW] >> 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 >> switch_ivr_originate() Can >> not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() >> Originate >> Failed. Cause: USER_NOT_REGISTERED >> >> 2009/6/2 Jason White >> >> Brad Tuan wrote: >> > I have tried >> > >> > >> > >> > > data="sofia/internal/$1%192.168.141.187"/> >> >> Change the % to an @ in the above. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/a8416abf/attachment-0001.html From plite2012 at gmail.com Tue Jun 2 20:46:16 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 2 Jun 2009 22:46:16 -0500 Subject: [Freeswitch-users] Sending fax on Windows: Any one has succeeded? Message-ID: Has anyone succeeded in sending fax on Windows with the following command line? originate sofia/gateway// &txfax(/path_to_fax_file) No matter how I specify that path (I even copied the file into the installation folder, C:\Program Files\FreeSWITCH), I always got "[ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file". Any hint would be highly appreciated! From brad.tuan at gmail.com Tue Jun 2 21:38:56 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 12:38:56 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: ......I've update my FS by SVN.......... but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" Is that right?? And the displayname is still "97730002"....... What i confused is why "97730002" ?? ( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) ) recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: ------------------------------------------------------------------------ INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 ;tag=124393762732run00 To: Contact: Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: From: 97719006 ;tag=124393762732run00 To: Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006 at 61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: ------------------------------------------------------------------------ INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" >;tag=F9rteQHjgS52m To: Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" >;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2009/6/3 Michael Collins > > > On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: > >> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? > > > Your best bet is to use SVN trunk. It is the most stable version available, > even more stable than the latest 1.0.4pre8 release candidate. Back up your > entire freeswitch folder in case there's an issue. Hopefully you're running > in Linux, so you could do: > mv /usr/local/freeswitch /usr/local/freeswitch.bak > > Then use the quick and dirty install from the wiki: > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > When the install is finished you will have a fresh copy of FS and a brand > new default configuration. You'll need to go back and enable and build any > modules you need that aren't done by default. You will also need to re-apply > any changes you made to the default configuration from your previous > install. Hopefully you didn't have to edit any of the files or maybe just a > few, like vars.xml. In any case, I recommend editing as few of the default > config files as possible. > > Let us know how it goes... > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/df2c45e6/attachment.html From msc at freeswitch.org Tue Jun 2 21:59:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 21:59:10 -0700 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> okay, you will need to use pastebin and post your configuration. anything you changed from the default config, especially in the dialplan, but also vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console loglevel 7") and also do the SIP trace. Make a few test calls and capture all the output. -MC On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > ......I've update my FS by SVN.......... > > but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" > > Is that right?? > > And the displayname is still "97730002"....... > > What i confused is why "97730002" ?? > > ( I have users from 97730000~97739999,but when I call them from 97710006 , > the display name is always "97730002"(it should be "97710006".....) ) > > recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: > ------------------------------------------------------------------------ > INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 > Via: SIP/2.0/UDP 61.61.162.130:5060 > ;branch=z9hG4bKrun12440031628377850000 > Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 > From: 97719006 ;tag=124393762732run00 > To: > Contact: > Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 > CSeq: 8500 INVITE > Max-Forwards: 68 > Content-Type: application/sdp > Content-Length: 237 > v=0 > o=169 0 0 IN IP4 61.61.162.130 > s=ots > c=IN IP4 61.61.162.130 > t=0 0 > m=audio 5158 RTP/AVP 18 8 0 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:18 G729/8000/1 > a=rtpmap:101 telephone-event/8000/1 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 > Via: SIP/2.0/UDP 61.61.162.130:5060 > ;branch=z9hG4bKrun12440031628377850000 > Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 > Record-Route: > From: 97719006 ;tag=124393762732run00 > To: > Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 > CSeq: 8500 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New > Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 > f34cc3da] > 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 977 > 19006->97730009 in context default > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 1 execute_extension::dx XML features > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 2 > record_session::C:\SipGo/recordings/97719006.2009-06-03-12 > -25-59.wav > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 3 execute_extension::cf XML features > 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New > Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 > ;rinstance=89358e5ea9aaa > 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] > 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 > switch_ivr_originate() Se > nding early media > 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring > SDP: > v=0 > o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 > s=FreeSWITCH > c=IN IP4 203.64.215.209 > t=0 0 > m=audio 17022 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() > Pre-Answer > sofia/internal/97719006 at 61.61.162.130:5060! > send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: > ------------------------------------------------------------------------ > INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e > SIP/2.0 > Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta > Max-Forwards: 67 > From: "Extension 97730002" > >;tag=F9rteQHjgS52m > To: > Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace > CSeq: 115883243 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-descrip > tion, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 449 > Remote-Party-ID: "Extension 97730002" > >;party=cal > ling;screen=yes;privacy=off > v=0 > o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 > 203.64.215.209 > s=FreeSWITCH > c=IN IP4 203.64.215.209 > t=0 0 > m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:115 G7221/32000 > a=fmtp:115 bitrate=48000 > a=rtpmap:107 G7221/16000 > a=fmtp:107 bitrate=32000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > 2009/6/3 Michael Collins > >> >> >> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >> >>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >> >> >> Your best bet is to use SVN trunk. It is the most stable version >> available, even more stable than the latest 1.0.4pre8 release candidate. >> Back up your entire freeswitch folder in case there's an issue. Hopefully >> you're running in Linux, so you could do: >> mv /usr/local/freeswitch /usr/local/freeswitch.bak >> >> Then use the quick and dirty install from the wiki: >> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >> >> When the install is finished you will have a fresh copy of FS and a brand >> new default configuration. You'll need to go back and enable and build any >> modules you need that aren't done by default. You will also need to re-apply >> any changes you made to the default configuration from your previous >> install. Hopefully you didn't have to edit any of the files or maybe just a >> few, like vars.xml. In any case, I recommend editing as few of the default >> config files as possible. >> >> Let us know how it goes... >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/1a1e43d3/attachment-0001.html From shaheryarkh at googlemail.com Tue Jun 2 22:09:22 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 3 Jun 2009 11:09:22 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Message-ID: I had to upgrade again svn revision to use this switch, but it works. Thank you. On Wed, Jun 3, 2009 at 1:12 AM, Keith Laaks wrote: > Hi, > > > > Try starting using the -nonat switch. > > > > Best Regards > > > > Keith > > > > *From:* Muhammad Shahzad [mailto:shaheryarkh at googlemail.com] > *Sent:* 02 June 2009 14:39 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch taking too long to start up > > > > Yes, this resolves the problem. > > Thank you. > > On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: > > Actually Brain mentioned that you can comment out switch_nat_init(); in > switch_core.c > > > > On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > > As I understand it, a new ?feature? was added over the weekend to > resolve NAT. If you?re firewall is not allowing ICMP then FS waits until it > times out. At this time there is no option to disable it. > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Muhammad Shahzad > *Sent:* 02 June 2009 11:40 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch taking too long to start up > > > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I am > using 32bit CentOS 5.3, "make current" command completes successfully > without any errors but when i start freeswitch it take considerable time > (roughly 90 - 120 seconds) to start up. During this time no message is > display on console. Once successfully started, it works fine. However, this > initial delay is really annoying. Is there anyway to reduce/remove this > delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/00066170/attachment.html From brad.tuan at gmail.com Tue Jun 2 22:26:42 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 13:26:42 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> Message-ID: I only change freeSWITCH\conf\dialplan\default.xml and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default 2009/6/3 Michael Collins > okay, you will need to use pastebin and post your configuration. anything > you changed from the default config, especially in the dialplan, but also > vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console > loglevel 7") and also do the SIP trace. Make a few test calls and capture > all the output. > > -MC > > > On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > >> ......I've update my FS by SVN.......... >> >> but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" >> >> Is that right?? >> >> And the displayname is still "97730002"....... >> >> What i confused is why "97730002" ?? >> >> ( I have users from 97730000~97739999,but when I call them from 97710006 , >> the display name is always "97730002"(it should be "97710006".....) ) >> >> recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 >> Record-Route: >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> From: 97719006 ;tag=124393762732run00 >> To: >> Contact: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> Max-Forwards: 68 >> Content-Type: application/sdp >> Content-Length: 237 >> v=0 >> o=169 0 0 IN IP4 61.61.162.130 >> s=ots >> c=IN IP4 61.61.162.130 >> t=0 0 >> m=audio 5158 RTP/AVP 18 8 0 101 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=rtpmap:101 telephone-event/8000/1 >> a=fmtp:101 0-15 >> >> ------------------------------------------------------------------------ >> send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> Record-Route: >> From: 97719006 ;tag=124393762732run00 >> To: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 >> f34cc3da] >> 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 977 >> 19006->97730009 in context default >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 1 execute_extension::dx XML features >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 2 >> record_session::C:\SipGo/recordings/97719006.2009-06-03-12 >> -25-59.wav >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 3 execute_extension::cf XML features >> 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 >> ;rinstance=89358e5ea9aaa >> 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] >> 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 >> switch_ivr_originate() Se >> nding early media >> 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring >> SDP: >> v=0 >> o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 17022 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() >> Pre-Answer >> sofia/internal/97719006 at 61.61.162.130:5060! >> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e >> SIP/2.0 >> Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta >> Max-Forwards: 67 >> From: "Extension 97730002" >> >;tag=F9rteQHjgS52m >> To: >> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace >> CSeq: 115883243 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-descrip >> tion, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 449 >> Remote-Party-ID: "Extension 97730002" >> >;party=cal >> ling;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 >> 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:115 G7221/32000 >> a=fmtp:115 bitrate=48000 >> a=rtpmap:107 G7221/16000 >> a=fmtp:107 bitrate=32000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> 2009/6/3 Michael Collins >> >>> >>> >>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >>> >>>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >>> >>> >>> Your best bet is to use SVN trunk. It is the most stable version >>> available, even more stable than the latest 1.0.4pre8 release candidate. >>> Back up your entire freeswitch folder in case there's an issue. Hopefully >>> you're running in Linux, so you could do: >>> mv /usr/local/freeswitch /usr/local/freeswitch.bak >>> >>> Then use the quick and dirty install from the wiki: >>> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >>> >>> When the install is finished you will have a fresh copy of FS and a brand >>> new default configuration. You'll need to go back and enable and build any >>> modules you need that aren't done by default. You will also need to re-apply >>> any changes you made to the default configuration from your previous >>> install. Hopefully you didn't have to edit any of the files or maybe just a >>> few, like vars.xml. In any case, I recommend editing as few of the default >>> config files as possible. >>> >>> Let us know how it goes... >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/3c0f510e/attachment-0001.html From woodydickson at gmail.com Tue Jun 2 23:22:11 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Wed, 3 Jun 2009 14:22:11 +0800 Subject: [Freeswitch-users] Set problem in dialplan Message-ID: Hello, I am getting a strange problem in my dialplan. After doing "SET", I want to use it in the next condition field. But then the value is not being set properly. Could someone please tell me what is wrong? Thanks, Woody Here is the dialplan: Here is the FS log. Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution->get-pin] continue=true Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [get-pin] ${destination_number}(117) =~ /^(.*)$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Action set(conference_id=111) Dialplan: sofia/internal/1001 at 192.168.1.101 Action set(is_moderator=true) Dialplan: sofia/internal/1001 at 192.168.1.101 Action info() Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution->conf] continue=false Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] ${is_moderator}() =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] ${is_moderator}() =~ /^false$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [conf] ${is_moderator}() =~ /^$/ break=always Dialplan: sofia/internal/1001 at 192.168.1.101 Action playback(/var/app/prompt/wav/bye.wav) Dialplan: sofia/internal/1001 at 192.168.1.101 Action hangup() -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/f6b6ec9b/attachment.html From mrene_lists at avgs.ca Tue Jun 2 23:26:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 02:26:32 -0400 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: Message-ID: <63B78532-2DF8-47E2-91CF-060AF95B8205@avgs.ca> Hi, FreeSWITCH decides what to execute first, the set application runs later (look a few lines later, you'll see lines beginning with EXECUTE, this is when it runs). If you need to use variables you've set in the DP, you need to use the transfer application to make it go back into routing state. Math On 3-Jun-09, at 2:22 AM, Woody Dickson wrote: > Hello, > > I am getting a strange problem in my dialplan. > > After doing "SET", I want to use it in the next condition field. > But then the value is not being set properly. > > Could someone please tell me what is wrong? > > Thanks, > Woody > > > Here is the dialplan: > > > break="never"> > > > > > > > > > > > > > > > > > > > > > > > > > > > Here is the FS log. > > Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution- > >get-pin] continue=true > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [get-pin] $ > {destination_number}(117) =~ /^(.*)$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Action > set(conference_id=111) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action > set(is_moderator=true) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action info() > Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution- > >conf] continue=false > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] $ > {is_moderator}() =~ /^true$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] $ > {is_moderator}() =~ /^false$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [conf] $ > {is_moderator}() =~ /^$/ break=always > Dialplan: sofia/internal/1001 at 192.168.1.101 Action playback(/var/app/ > prompt/wav/bye.wav) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action hangup() > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/1efcaa0a/attachment.html From jim at evolutiontel.net Tue Jun 2 23:32:49 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 3 Jun 2009 16:32:49 +1000 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <302665.69883.qm@web34301.mail.mud.yahoo.com> References: <302665.69883.qm@web34301.mail.mud.yahoo.com> Message-ID: Fernando, Try setting 'inbound-late-negotiation' in your SIP Profile. This will allow the call to hit the dialplan where you can set proxy_media. This also assumes you have bypass_media set to false in your dialplan. Alternatively I beleive you can set "inbound-proxy-media" in the SIP Profile and this will do the same thing. Regards, Jim On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL wrote: > > Dear, > > I can't solve my problem, i was try with: > > > > > and: > > in freeswitch.xml > > But receive the same log: > > http://pastebin.freeswitch.org/9204 > > Anyone help me. > > Fernando > > --- On Mon, 6/1/09, FERNANDO VILLARROEL wrote: > >> From: FERNANDO VILLARROEL >> Subject: Re: [Freeswitch-users] Passthru mode >> To: freeswitch-users at lists.freeswitch.org >> Date: Monday, June 1, 2009, 8:10 PM >> >> Hello i was try with: >> >> >> > data="sofia/gateway/ubb/$1$2$3"/> >> >> This is the log on FS_CLI: >> >> http://pastebin.freeswitch.org/9204 >> >> Fernando >> >> --- On Mon, 6/1/09, Michael Collins >> wrote: >> >> > From: Michael Collins >> > Subject: Re: [Freeswitch-users] Passthru mode >> > To: freeswitch-users at lists.freeswitch.org >> > Date: Monday, June 1, 2009, 7:41 PM >> > >> > >> > On Mon, Jun 1, 2009 at 3:20 PM, >> > FERNANDO VILLARROEL >> > wrote: >> > >> > >> > >> > Hello the dial plan: >> > >> > >> > >> > > > data="sofia/gateway/ubb/$1$2$3"/> >> > >> > >> > >> > This i setup from Wikipbx. >> > What about this in the dialplan? >> > > > data="proxy_media=true"/> >> > Or alternatively this in the SIP profile? >> > >> > > > value="true"/> >> > >> > I just want to make sure you're actually telling FS >> to >> > use proxy media. If I may make a suggestion: use >> pastebin.freeswitch.org >> > and pastebin the entire extension in the dialplan as >> well as >> > a complete debug log of the call from the FS CLI. >> Please see >> > this page for some handy tips on gathering information >> for >> > troubleshooting: >> > >> > http://wiki.freeswitch.org/wiki/Reporting_Bugs >> > >> > -MC >> > >> > >> > >> > -----Inline Attachment Follows----- >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mrene_lists at avgs.ca Tue Jun 2 23:33:45 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 02:33:45 -0400 Subject: [Freeswitch-users] Passthru mode In-Reply-To: References: <302665.69883.qm@web34301.mail.mud.yahoo.com> Message-ID: <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> On 3-Jun-09, at 2:32 AM, Jim Burke wrote: > Fernando, > > Try setting 'inbound-late-negotiation' in your SIP Profile. This will > allow the call to hit the dialplan where you can set proxy_media. > This also assumes you have bypass_media set to false in your dialplan. > > Alternatively I beleive you can set "inbound-proxy-media" in the SIP > Profile and this will do the same thing. But you still need late negotiation for that to work, so in both cases you need to fix that :D Math > > > Regards, > Jim > > On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL > wrote: >> >> Dear, >> >> I can't solve my problem, i was try with: >> >> >> >> >> and: >> >> in freeswitch.xml >> >> But receive the same log: >> >> http://pastebin.freeswitch.org/9204 >> >> Anyone help me. >> >> Fernando >> >> --- On Mon, 6/1/09, FERNANDO VILLARROEL >> wrote: >> >>> From: FERNANDO VILLARROEL >>> Subject: Re: [Freeswitch-users] Passthru mode >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Monday, June 1, 2009, 8:10 PM >>> >>> Hello i was try with: >>> >>> >>> >> data="sofia/gateway/ubb/$1$2$3"/> >>> >>> This is the log on FS_CLI: >>> >>> http://pastebin.freeswitch.org/9204 >>> >>> Fernando >>> >>> --- On Mon, 6/1/09, Michael Collins >>> wrote: >>> >>>> From: Michael Collins >>>> Subject: Re: [Freeswitch-users] Passthru mode >>>> To: freeswitch-users at lists.freeswitch.org >>>> Date: Monday, June 1, 2009, 7:41 PM >>>> >>>> >>>> On Mon, Jun 1, 2009 at 3:20 PM, >>>> FERNANDO VILLARROEL >>>> wrote: >>>> >>>> >>>> >>>> Hello the dial plan: >>>> >>>> >>>> >>>> >>> data="sofia/gateway/ubb/$1$2$3"/> >>>> >>>> >>>> >>>> This i setup from Wikipbx. >>>> What about this in the dialplan? >>>> >>> data="proxy_media=true"/> >>>> Or alternatively this in the SIP profile? >>>> >>>> >>> value="true"/> >>>> >>>> I just want to make sure you're actually telling FS >>> to >>>> use proxy media. If I may make a suggestion: use >>> pastebin.freeswitch.org >>>> and pastebin the entire extension in the dialplan as >>> well as >>>> a complete debug log of the call from the FS CLI. >>> Please see >>>> this page for some handy tips on gathering information >>> for >>>> troubleshooting: >>>> >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> -MC >>>> >>>> >>>> >>>> -----Inline Attachment Follows----- >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Tue Jun 2 23:39:09 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 3 Jun 2009 16:39:09 +1000 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: Message-ID: <20090603063909.GA19487@jdc.jasonjgw.net> Woody Dickson wrote: > I am getting a strange problem in my dialplan. > > After doing "SET", I want to use it in the next condition field. But then > the value is not being set properly. When parsing the dial plan, FreeSWITCH tests all of the conditions, then builds a linked list of actions to execute. Once this is done, the actions are executed, in order. This is why you can't simply set a variable in one extension and test it in the condition of a later extension. From bruce.mcalister at blueface.ie Wed Jun 3 00:16:01 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 08:16:01 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" Message-ID: <4A262331.2080104@blueface.ie> Hi All, I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the build fails with the following error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" cc: acomp failed for locks/unix/thread_mutex.c make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 I have the JDS-CBE build environment setup as recommended on the wiki with Sun Studio 12 compiler suite installed as well. I have tried building using Sun's compiler and GNU compiler but I get the same error message. I just recently tried "bootstrap.sh" prior to "configure", "make" but the error is still the same. Would someone have any suggestions for me to try to get around this? Thanks Bruce From bruce.mcalister at blueface.ie Wed Jun 3 02:47:43 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 10:47:43 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <4A262331.2080104@blueface.ie> References: <4A262331.2080104@blueface.ie> Message-ID: <4A2646BF.7090607@blueface.ie> Hi All, I get past this initial error if I change my C compiler from "usr/bin/cc" to "/usr/bin/c99". After changing the above, the compilation goes further, but I am now faced with a different error: --- /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/c99 -m32 -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o threadproc/unix/signals.lo -c threadproc/unix/signals.c && touch threadproc/unix/signals.lo "threadproc/unix/signals.c", line 48: warning: implicit function declaration: kill "threadproc/unix/signals.c", line 76: incomplete struct/union/enum sigaction: act "threadproc/unix/signals.c", line 78: undefined struct/union member: sa_handler "threadproc/unix/signals.c", line 78: warning: improper pointer/integer combination: op "=" "threadproc/unix/signals.c", line 79: warning: implicit function declaration: sigemptyset "threadproc/unix/signals.c", line 79: undefined struct/union member: sa_mask "threadproc/unix/signals.c", line 80: undefined struct/union member: sa_flags "threadproc/unix/signals.c", line 103: warning: implicit function declaration: sigaction "threadproc/unix/signals.c", line 105: improper member use: sa_handler "threadproc/unix/signals.c", line 105: warning: improper pointer/integer combination: op "=" "threadproc/unix/signals.c", line 277: warning: implicit function declaration: sigdelset "threadproc/unix/signals.c", line 327: warning: implicit function declaration: sigfillset "threadproc/unix/signals.c", line 424: warning: implicit function declaration: pthread_sigmask "threadproc/unix/signals.c", line 443: warning: implicit function declaration: sigaddset c99: acomp failed for threadproc/unix/signals.c make[2]: *** [threadproc/unix/signals.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 --- Any hints towards a solution would be appreciated. Thanks Bruce Bruce McAlister wrote: > Hi All, > > I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the > build fails with the following error: > > /bin/bash > /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool > --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 > -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE > -I./include > -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix > -I./include/arch/unix > -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include > -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch > locks/unix/thread_mutex.lo > "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or > options invalid; UNIX 03 and POSIX.1-2001 applications require the > use of c99" > cc: acomp failed for locks/unix/thread_mutex.c > make[2]: *** [locks/unix/thread_mutex.lo] Error 1 > make[2]: Leaving directory > `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory > `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' > make: *** [libs/apr/libapr-1.la] Error 2 > From jason at jasonjgw.net Wed Jun 3 03:17:46 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 3 Jun 2009 20:17:46 +1000 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <4A2646BF.7090607@blueface.ie> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> Message-ID: <20090603101746.GA3854@jdc.jasonjgw.net> Bruce McAlister wrote: > I get past this initial error if I change my C compiler from > "usr/bin/cc" to "/usr/bin/c99". > > After changing the above, the compilation goes further, but I am now > faced with a different error: Have you tried compiling with gcc? I would also suggest starting the build procedure from the beginning. From brad.tuan at gmail.com Wed Jun 3 04:16:16 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 19:16:16 +0800 Subject: [Freeswitch-users] Little problem of group(callgroup) Message-ID: What is the ${callgroup} mean?? Is this?? >> Or this?? >>81+[group] - Add this extension to calling group #[group] (can be two digits 00-99). A beep tone confirms the function worked. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/40ed8ef0/attachment.html From rdenert at tng.de Wed Jun 3 04:18:25 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 13:18:25 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <28083808.277891244027846976.JavaMail.root@zimbra.tng.de> Message-ID: <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> Hello, I have a question about the RTP stream. I made two calls with different devices. One had no problems, the other call made some difficulties. I put an extraction of my traces in attachment. (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) The first one had no problem. When I called the freeSWITCH I had a bidirectional RTP stream. I heard the announcement of the server. I could transmit digits from my telephone to the freeSWITCH which were verified by the machine. (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 phone) The second call had only a oneway audio stream. When I called the freeSWITCH I heard no announcement of the server which should be actually played. There was only a background noise. I made traces from this example. My suggestion is: In packet 378 the freeSWITCH server wants to sent RTP packets to an suspicious IP. There are over 5 packets in number. Not till then there is the correct destination IP (see packet 386). But this could be the fact that the freeSWITCH produces an error. Does anybody have an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. 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Name: traces_from 1616_to_freeSWITCH.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/19fd92f1/attachment.txt From bruce.mcalister at blueface.ie Wed Jun 3 04:36:05 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 12:36:05 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <20090603101746.GA3854@jdc.jasonjgw.net> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> <20090603101746.GA3854@jdc.jasonjgw.net> Message-ID: <4A266025.1070505@blueface.ie> Hi Jason, If I try to compile with GCC, then I am faced with the original problem where the error returns saying I need to use a c99 compatible compiler, here is the specific error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/sfw/bin/gcc -m32 -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo In file included from /usr/sfw/lib/gcc/i386-pc-solaris2.10/3.4.3/include/sys/types.h:27, from ./include/apr.h:113, from /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix/apr_arch_thread_mutex.h:24, from locks/unix/thread_mutex.c:17: /usr/include/sys/feature_tests.h:336:2: #error "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 In all cases I have started the build from the beginning, whereby I remove and re-extract the 1.0.4pre8 tarball. I've tried with just a configure and also a bootstrap/configure, but I end up with the same error (except when I change the compiler to Sun Studio 12's c99). Is GCC 3.4.3 too old to use to build this version of freeswitch? Thanks Bruce Jason White wrote: > Bruce McAlister wrote: >> I get past this initial error if I change my C compiler from >> "usr/bin/cc" to "/usr/bin/c99". >> >> After changing the above, the compilation goes further, but I am now >> faced with a different error: > > Have you tried compiling with gcc? I would also suggest starting the build > procedure from the beginning. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brad.tuan at gmail.com Wed Jun 3 06:04:33 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 21:04:33 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> Message-ID: I only change freeSWITCH\conf\dialplan\default.xml and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default 2009/6/3 Michael Collins > okay, you will need to use pastebin and post your configuration. anything > you changed from the default config, especially in the dialplan, but also > vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console > loglevel 7") and also do the SIP trace. Make a few test calls and capture > all the output. > > -MC > > > On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > >> ......I've update my FS by SVN.......... >> >> but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" >> >> Is that right?? >> >> And the displayname is still "97730002"....... >> >> What i confused is why "97730002" ?? >> >> ( I have users from 97730000~97739999,but when I call them from 97710006 , >> the display name is always "97730002"(it should be "97710006".....) ) >> >> recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 >> Record-Route: >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> From: 97719006 ;tag=124393762732run00 >> To: >> Contact: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> Max-Forwards: 68 >> Content-Type: application/sdp >> Content-Length: 237 >> v=0 >> o=169 0 0 IN IP4 61.61.162.130 >> s=ots >> c=IN IP4 61.61.162.130 >> t=0 0 >> m=audio 5158 RTP/AVP 18 8 0 101 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=rtpmap:101 telephone-event/8000/1 >> a=fmtp:101 0-15 >> >> ------------------------------------------------------------------------ >> send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> Record-Route: >> From: 97719006 ;tag=124393762732run00 >> To: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 >> f34cc3da] >> 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 977 >> 19006->97730009 in context default >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 1 execute_extension::dx XML features >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 2 >> record_session::C:\SipGo/recordings/97719006.2009-06-03-12 >> -25-59.wav >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 3 execute_extension::cf XML features >> 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 >> ;rinstance=89358e5ea9aaa >> 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] >> 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 >> switch_ivr_originate() Se >> nding early media >> 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring >> SDP: >> v=0 >> o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 17022 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() >> Pre-Answer >> sofia/internal/97719006 at 61.61.162.130:5060! >> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e >> SIP/2.0 >> Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta >> Max-Forwards: 67 >> From: "Extension 97730002" >> >;tag=F9rteQHjgS52m >> To: >> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace >> CSeq: 115883243 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-descrip >> tion, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 449 >> Remote-Party-ID: "Extension 97730002" >> >;party=cal >> ling;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 >> 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:115 G7221/32000 >> a=fmtp:115 bitrate=48000 >> a=rtpmap:107 G7221/16000 >> a=fmtp:107 bitrate=32000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> 2009/6/3 Michael Collins >> >>> >>> >>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >>> >>>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >>> >>> >>> Your best bet is to use SVN trunk. It is the most stable version >>> available, even more stable than the latest 1.0.4pre8 release candidate. >>> Back up your entire freeswitch folder in case there's an issue. Hopefully >>> you're running in Linux, so you could do: >>> mv /usr/local/freeswitch /usr/local/freeswitch.bak >>> >>> Then use the quick and dirty install from the wiki: >>> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >>> >>> When the install is finished you will have a fresh copy of FS and a brand >>> new default configuration. You'll need to go back and enable and build any >>> modules you need that aren't done by default. You will also need to re-apply >>> any changes you made to the default configuration from your previous >>> install. Hopefully you didn't have to edit any of the files or maybe just a >>> few, like vars.xml. In any case, I recommend editing as few of the default >>> config files as possible. >>> >>> Let us know how it goes... >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/49e162f1/attachment.html From brad.tuan at gmail.com Wed Jun 3 06:06:03 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 21:06:03 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: I was updated my FS and rebuilt it. It works........ But when User2(FS2) accept the call from User1(FS1) , User2(FS2) display "call established",but User1(FS1) still display "calling". Why?? (I think maybe that I need to do some setting on FS2.) 2009/6/2 Brad Tuan > FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 > > These two FS are in the same LAN. > > I just try to pass one sip call from one FS to another. > > If it works, next is FS1( PublicIP ) to FS2( PublicIP ). > > 2009/6/2 Ken Rice > > Dumb question... Is 187 the local fs machine? You should have the IP >> address of the remote FS machine >> >> >> ------------------------------ >> *From: *Brad Tuan >> *Reply-To: * >> *Date: *Tue, 2 Jun 2009 17:00:29 +0800 >> *To: * >> *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to >> another FS ?? >> >> >> the same message........ >> >> 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 100 >> 1->97710001 in context default >> 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() >> Cannot l >> ocate registered user 97710001 at 192.168.141.187 at internal >> 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() >> Close Cha >> nnel N/A [CS_NEW] >> 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 >> switch_ivr_originate() Can >> not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() >> Originate >> Failed. Cause: USER_NOT_REGISTERED >> >> 2009/6/2 Jason White >> >> Brad Tuan wrote: >> > I have tried >> > >> > >> > >> > > data="sofia/internal/$1%192.168.141.187"/> >> >> Change the % to an @ in the above. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/e0c80105/attachment-0001.html From brian at freeswitch.org Wed Jun 3 06:18:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 08:18:14 -0500 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Message-ID: <59F86087-9012-40D0-87F7-4721E49FAA7F@freeswitch.org> You shouldn't have to use the switch anymore. That is unless you just wanna skip that check. /b On Jun 3, 2009, at 12:09 AM, Muhammad Shahzad wrote: > I had to upgrade again svn revision to use this switch, but it works. > > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/26ff7cca/attachment.html From anthony.minessale at gmail.com Wed Jun 3 06:18:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:18:50 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> References: <28083808.277891244027846976.JavaMail.root@zimbra.tng.de> <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> Message-ID: <191c3a030906030618g2f6bec95lcbfe966565aefd7a@mail.gmail.com> This is called RTP Auto Adjust This occurs when the SDP of the other side sends FS the wrong media IP in the SDP If FS gets packets from some other place besides where it thinks its supposed to send packets in a window of the first 10 packets repeatedly, then it auto adjusts the destination, fixing the problem. If you look at your FreeSWITCH console when you make this call it's likely you see a message about RTP auto adjusting the IP. On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert wrote: > Hello, > > I have a question about the RTP stream. I made two calls with different > devices. One had no problems, the other call made some difficulties. I put > an extraction of my traces in attachment. > > (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) > The first one had no problem. When I called the freeSWITCH I had a > bidirectional RTP stream. I heard the announcement of the server. I could > transmit digits from my telephone to the freeSWITCH which were verified by > the machine. > > (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 > phone) > The second call had only a oneway audio stream. When I called the > freeSWITCH I heard no announcement of the server which should be actually > played. There was only a background noise. I made traces from this example. > > My suggestion is: > In packet 378 the freeSWITCH server wants to sent RTP packets to an > suspicious IP. There are over 5 packets in number. Not till then there is > the correct destination IP (see packet 386). But this could be the fact that > the freeSWITCH produces an error. > > Does anybody have an idea? > > Greetz > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/3e86bfab/attachment.html From anthony.minessale at gmail.com Wed Jun 3 06:28:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:28:02 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> also press f8 before you take the console log to get the debugging info and paste the resulting trace in http://pastebin.freeswitch.org rather than right in the email -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/222da725/attachment.html From anthony.minessale at gmail.com Wed Jun 3 06:39:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:39:21 -0500 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> References: <302665.69883.qm@web34301.mail.mud.yahoo.com> <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> Message-ID: <191c3a030906030639wfa33c72qe265631699a00fb9@mail.gmail.com> I am pretty sure inbound-proxy-media forces late-negotation iirc. On Wed, Jun 3, 2009 at 1:33 AM, Mathieu Rene wrote: > > On 3-Jun-09, at 2:32 AM, Jim Burke wrote: > > > Fernando, > > > > Try setting 'inbound-late-negotiation' in your SIP Profile. This will > > allow the call to hit the dialplan where you can set proxy_media. > > This also assumes you have bypass_media set to false in your dialplan. > > > > Alternatively I beleive you can set "inbound-proxy-media" in the SIP > > Profile and this will do the same thing. > > But you still need late negotiation for that to work, so in both cases > you need to fix that :D > > Math > > > > > > > Regards, > > Jim > > > > On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL > > wrote: > >> > >> Dear, > >> > >> I can't solve my problem, i was try with: > >> > >> > >> > >> > >> and: > >> > >> in freeswitch.xml > >> > >> But receive the same log: > >> > >> http://pastebin.freeswitch.org/9204 > >> > >> Anyone help me. > >> > >> Fernando > >> > >> --- On Mon, 6/1/09, FERNANDO VILLARROEL > >> wrote: > >> > >>> From: FERNANDO VILLARROEL > >>> Subject: Re: [Freeswitch-users] Passthru mode > >>> To: freeswitch-users at lists.freeswitch.org > >>> Date: Monday, June 1, 2009, 8:10 PM > >>> > >>> Hello i was try with: > >>> > >>> > >>> >>> data="sofia/gateway/ubb/$1$2$3"/> > >>> > >>> This is the log on FS_CLI: > >>> > >>> http://pastebin.freeswitch.org/9204 > >>> > >>> Fernando > >>> > >>> --- On Mon, 6/1/09, Michael Collins > >>> wrote: > >>> > >>>> From: Michael Collins > >>>> Subject: Re: [Freeswitch-users] Passthru mode > >>>> To: freeswitch-users at lists.freeswitch.org > >>>> Date: Monday, June 1, 2009, 7:41 PM > >>>> > >>>> > >>>> On Mon, Jun 1, 2009 at 3:20 PM, > >>>> FERNANDO VILLARROEL > >>>> wrote: > >>>> > >>>> > >>>> > >>>> Hello the dial plan: > >>>> > >>>> > >>>> > >>>> >>>> data="sofia/gateway/ubb/$1$2$3"/> > >>>> > >>>> > >>>> > >>>> This i setup from Wikipbx. > >>>> What about this in the dialplan? > >>>> >>>> data="proxy_media=true"/> > >>>> Or alternatively this in the SIP profile? > >>>> > >>>> >>>> value="true"/> > >>>> > >>>> I just want to make sure you're actually telling FS > >>> to > >>>> use proxy media. If I may make a suggestion: use > >>> pastebin.freeswitch.org > >>>> and pastebin the entire extension in the dialplan as > >>> well as > >>>> a complete debug log of the call from the FS CLI. > >>> Please see > >>>> this page for some handy tips on gathering information > >>> for > >>>> troubleshooting: > >>>> > >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs > >>>> > >>>> -MC > >>>> > >>>> > >>>> > >>>> -----Inline Attachment Follows----- > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/7551d211/attachment-0001.html From asannucci at gmail.com Wed Jun 3 08:03:51 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:03:51 +0200 Subject: [Freeswitch-users] Error sofia_reg_c Message-ID: <7D3E2C7F9090451F9809F828584780B1@voztovoice> This is my problem: i have two gateway configured on FS that working fine (registered) when i start FS on the fs_cli I receive this message [ERR] sofia_reg.c:1499 sofia_reg_handle_sip_r_challenge() No Matching gateway found What means this error? Thank you. Best regards From brian at freeswitch.org Wed Jun 3 08:09:23 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:09:23 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <7D3E2C7F9090451F9809F828584780B1@voztovoice> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> Message-ID: <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> It means the far side send us a 407 and we couldn't match it to any gateway on your system to answer the challenge so we have no choice but to fail the call. /b On Jun 3, 2009, at 10:03 AM, bakko wrote: > This is my problem: > > i have two gateway configured on FS that working fine (registered) > > when i start FS on the fs_cli I receive this message > > [ERR] sofia_reg.c:1499 sofia_reg_handle_sip_r_challenge() No Matching > gateway found > > What means this error? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/5bb77b80/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 04:52:16 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 04:52:16 -0700 Subject: [Freeswitch-users] Softphone configuration Message-ID: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> Hi, I've just managed to get FreeSWITCH installed. I'm using the default config files which works fine with X-Lite. The problem is, I can't use any other softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is having issues with my Netgear router (it causes it to continually restart ... there are other posts about this elsewhere and the solutions aren't working.) The only other relevant thing I can think of to add to this topic is that X-Lite, on its first registration always gives me a timeout error, and then successfully registers. After a few minutes, the router reboots. Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/04970b99/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 06:20:53 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 06:20:53 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> Message-ID: <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> Hi, I've just managed to get FreeSWITCH installed. I'm using the default config files which works fine with X-Lite. The problem is, I can't use any other softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is having issues with my Netgear router (it causes it to continually restart ... there are other posts about this elsewhere and the solutions aren't working.) The only other relevant thing I can think of to add to this topic is that X-Lite, on its first registration always gives me a timeout error, and then successfully registers. After a few minutes, the router reboots. Matt PS. I apologize if this posts twice - I seemed to have an issue with my mail client and I don't think it sent the first time. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/198c78e7/attachment.html From asannucci at gmail.com Wed Jun 3 08:16:26 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:16:26 +0200 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: Ok. But why i receive a 407 response if no call in progress or active. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/710307be/attachment.html From brian at freeswitch.org Wed Jun 3 08:23:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:23:18 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <5A7C2AFB-3E3B-4212-A81B-0830A1D629A1@freeswitch.org> Ok again just guessing because you failed to provide any info but the one line in your first email... could be a bad gateway name? no clue since you guessed we only needed to see the ONE line. I would put the log on pastebin join #freeswitch on IRC and ask.. this email stuff is too slow. /b On Jun 3, 2009, at 10:16 AM, bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. > > Regards. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/54b6f223/attachment.html From brian at freeswitch.org Wed Jun 3 08:18:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:18:00 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> Message-ID: <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> I'm going to have to guess that you're doing this all on the same machine? /b On Jun 3, 2009, at 8:20 AM, Matthew Lockwood wrote: > Hi, > > I've just managed to get FreeSWITCH installed. I'm using the default > config files which works fine with X-Lite. The problem is, I can't > use any other softphone other than X-Lite. i've tried YakaPhone, > Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say > this is a problem, but X-Lite is having issues with my Netgear > router (it causes it to continually restart ... there are other > posts about this elsewhere and the solutions aren't working.) > > The only other relevant thing I can think of to add to this topic is > that X-Lite, on its first registration always gives me a timeout > error, and then successfully registers. After a few minutes, the > router reboots. > > Matt > > PS. I apologize if this posts twice - I seemed to have an issue with > my mail client and I don't think it sent the first time. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0728eb15/attachment-0001.html From intralanman at freeswitch.org Wed Jun 3 08:23:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 11:23:58 -0400 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <4A26958E.4040503@freeswitch.org> bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. REMEMBER: ngrep is your friend. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/bb88cc34/attachment.html From rdenert at tng.de Wed Jun 3 08:29:45 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 17:29:45 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <17870203.282961244042976688.JavaMail.root@zimbra.tng.de> Message-ID: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> How can I avoid the problem? It seems that RTP auto adjust generates the error. Maybe I can deactivate RTP auto adjust but I suspect that freeSWITCH doesn't find the right media IP. Do you have any other solution? Greetz ----- Urspr?ngliche Mail ----- Von: "Anthony Minessale" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 3. Juni 2009 15:18:50 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Insufficient RTP stream This is called RTP Auto Adjust This occurs when the SDP of the other side sends FS the wrong media IP in the SDP If FS gets packets from some other place besides where it thinks its supposed to send packets in a window of the first 10 packets repeatedly, then it auto adjusts the destination, fixing the problem. If you look at your FreeSWITCH console when you make this call it's likely you see a message about RTP auto adjusting the IP. On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have a question about the RTP stream. I made two calls with different devices. One had no problems, the other call made some difficulties. I put an extraction of my traces in attachment. (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) The first one had no problem. When I called the freeSWITCH I had a bidirectional RTP stream. I heard the announcement of the server. I could transmit digits from my telephone to the freeSWITCH which were verified by the machine. (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 phone) The second call had only a oneway audio stream. When I called the freeSWITCH I heard no announcement of the server which should be actually played. There was only a background noise. I made traces from this example. My suggestion is: In packet 378 the freeSWITCH server wants to sent RTP packets to an suspicious IP. There are over 5 packets in number. Not till then there is the correct destination IP (see packet 386). But this could be the fact that the freeSWITCH produces an error. Does anybody have an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From matthew.lockwood at gmail.com Wed Jun 3 08:34:06 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 08:34:06 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> Message-ID: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> No, FS is installed on a VPS. I'm just connecting through my cable connection. M On Wed, Jun 3, 2009 at 8:18 AM, Brian West wrote: > I'm going to have to guess that you're doing this all on the same machine? > /b > > On Jun 3, 2009, at 8:20 AM, Matthew Lockwood wrote: > > Hi, > > I've just managed to get FreeSWITCH installed. I'm using the default config > files which works fine with X-Lite. The problem is, I can't use any other > softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, > QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is > having issues with my Netgear router (it causes it to continually restart > ... there are other posts about this elsewhere and the solutions aren't > working.) > > The only other relevant thing I can think of to add to this topic is that > X-Lite, on its first registration always gives me a timeout error, and then > successfully registers. After a few minutes, the router reboots. > > Matt > > PS. I apologize if this posts twice - I seemed to have an issue with my > mail client and I don't think it sent the first time. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/53f52dfa/attachment.html From rdenert at tng.de Wed Jun 3 08:35:17 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 17:35:17 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <31324436.283071244043300883.JavaMail.root@zimbra.tng.de> Message-ID: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> Is there a possibility to activate all DTMF detection modes (in-band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same extension of the dialplan? Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 21:29:19 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. Actually, that's very interesting. The start_dtmf app looks for in-band DTMFs, so if you have to have that in order to hear digits from the other end then the other end most definitely isn't doing RFC2833. For the sake of testing, try sending in-band and see how the other end reacts. Might want to check this out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate My guess is that the equipment along the way is futzing with things. FreeSWITCH does break easily but it does find bugs in other VoIP systems that it talks to... :) -MC The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" < msc at freeswitch.org > An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From intralanman at freeswitch.org Wed Jun 3 08:36:42 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 11:36:42 -0400 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: <4A26988A.1040308@freeswitch.org> Matthew Lockwood wrote: > No, FS is installed on a VPS. I'm just connecting through my cable > connection. > SPI or SIP ALG on the router? -Ray From brian at freeswitch.org Wed Jun 3 08:38:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:38:45 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: I'm guessing NAT problem, Not sure those other phones do STUN and traverse nat properly. /b On Jun 3, 2009, at 10:34 AM, Matthew Lockwood wrote: > No, FS is installed on a VPS. I'm just connecting through my cable > connection. > > M Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/5f1b436e/attachment-0001.html From brian at freeswitch.org Wed Jun 3 08:39:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:39:42 -0500 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> References: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> Message-ID: NO. You should only have one at a time. Its impossible to handle the scenario where you receive say on 2833 and then end up sending an info packet, 2833 packet and the inband of the DTMF... triple DTMF. :) /b On Jun 3, 2009, at 10:35 AM, Rudolf Denert wrote: > Is there a possibility to activate all DTMF detection modes (in- > band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same > extension of the dialplan? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/8e5be88c/attachment.html From dujinfang at gmail.com Wed Jun 3 08:41:46 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 23:41:46 +0800 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> I had this problem when I gateway to an asterisk box. Each time I call to asterisk through that gateway got a 407 and fail. Never figured out why but guess it's non-proper configuration of Asterisk. On Jun 3, 2009, at 11:16 PM, bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. > > Regards. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/d17d9a1d/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 08:42:30 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 08:42:30 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <4A26988A.1040308@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> Message-ID: <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> SPI disabled. There's no obvious option to disable SIP ALG, but NAT filtering was changed from secure to open. Problem still persists. On Wed, Jun 3, 2009 at 8:36 AM, Raymond Chandler wrote: > Matthew Lockwood wrote: > > No, FS is installed on a VPS. I'm just connecting through my cable > > connection. > > > SPI or SIP ALG on the router? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/eda81d32/attachment.html From dujinfang at gmail.com Wed Jun 3 08:43:08 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 23:43:08 +0800 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> References: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> Message-ID: <413F2852-3088-4035-BA26-B84C5BFE51EA@gmail.com> have you tried this? http://wiki.freeswitch.org/wiki/Channel_Variables#disable_rtp_auto_adjust On Jun 3, 2009, at 11:29 PM, Rudolf Denert wrote: > How can I avoid the problem? It seems that RTP auto adjust generates > the error. Maybe I can deactivate RTP auto adjust but I suspect that > freeSWITCH doesn't find the right media IP. > > Do you have any other solution? > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Anthony Minessale" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Mittwoch, 3. Juni 2009 15:18:50 GMT +01:00 Amsterdam/ > Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Insufficient RTP stream > > > This is called RTP Auto Adjust > This occurs when the SDP of the other side sends FS the wrong media > IP in the SDP > If FS gets packets from some other place besides where it thinks its > supposed to send packets in a window of the first 10 packets > repeatedly, > then it auto adjusts the destination, fixing the problem. If you > look at your FreeSWITCH console when you make this call it's likely > you see a message about RTP auto adjusting the IP. > > > > On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert < rdenert at tng.de > > wrote: > > > Hello, > > I have a question about the RTP stream. I made two calls with > different devices. One had no problems, the other call made some > difficulties. I put an extraction of my traces in attachment. > > (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 > phone) > The first one had no problem. When I called the freeSWITCH I had a > bidirectional RTP stream. I heard the announcement of the server. I > could transmit digits from my telephone to the freeSWITCH which were > verified by the machine. > > (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset > 5020 phone) > The second call had only a oneway audio stream. When I called the > freeSWITCH I heard no announcement of the server which should be > actually played. There was only a background noise. I made traces > from this example. > > My suggestion is: > In packet 378 the freeSWITCH server wants to sent RTP packets to an > suspicious IP. There are over 5 packets in number. Not till then > there is the correct destination IP (see packet 386). But this could > be the fact that the freeSWITCH produces an error. > > Does anybody have an idea? > > Greetz > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. > Any > unauthorized copying, disclosure or distribution of the material in > this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. > Any > unauthorized copying, disclosure or distribution of the material in > this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From asannucci at gmail.com Wed Jun 3 08:41:54 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:41:54 +0200 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com><415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com><7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: Have you open the necesary ports in the VPS firewall? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/57d8716b/attachment.html From brian at freeswitch.org Wed Jun 3 08:45:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:45:44 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> Message-ID: <1977D7D9-E777-40DC-A24D-AB4894471E6F@freeswitch.org> Its because its challenging you and you can't answer the challenge... its saying HEY YOU give me a user/pass ... and you can't answer that so it fails. /b On Jun 3, 2009, at 10:41 AM, dujinfang wrote: > I had this problem when I gateway to an asterisk box. Each time I > call to asterisk through that gateway got a 407 and fail. Never > figured out why but guess it's non-proper configuration of Asterisk. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/1626d3d0/attachment-0001.html From brian at freeswitch.org Wed Jun 3 08:46:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:46:14 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> References: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> Message-ID: <17D2F89B-F6ED-4DD7-9C1F-FEDBD03F7D67@freeswitch.org> Without a sip trace its hard to tell. /b On Jun 3, 2009, at 10:29 AM, Rudolf Denert wrote: > How can I avoid the problem? It seems that RTP auto adjust generates > the error. Maybe I can deactivate RTP auto adjust but I suspect that > freeSWITCH doesn't find the right media IP. > > Do you have any other solution? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/ac802066/attachment.html From brian at freeswitch.org Wed Jun 3 08:47:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:47:30 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> Message-ID: Thats not the problem.. Phone registers to freeswitch has 192.168.0.100 in the sip packet... FS sends a challenge back to 192.168.0.100 which obviously will FAIL cuz that network isn't anywhere near the VPS box... Go to the FreeSWITCH box and type "sofia profile internal siptrace on" and I'll suspect you see this exact behavior. /b On Jun 3, 2009, at 10:42 AM, Matthew Lockwood wrote: > SPI disabled. There's no obvious option to disable SIP ALG, but NAT > filtering was changed from secure to open. Problem still persists. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/95effb84/attachment.html From asannucci at gmail.com Wed Jun 3 09:24:35 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 18:24:35 +0200 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice><96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> Message-ID: This is not my problem. I have conected the asterisk to FS and i can call from FS to Asterisk without problem. I do some tests to call a international number using a sip provider that is registered on the asterisk pbx and work. Actualy i can: call from any FS extension any Asterisk extension call from any asterisk extension any fs extension I tried to disable the asterisk gateway configuration but not resolve the issue. Maybe the problem is with the other gateway. I have to investigate :) Sorry for my very bad english :) Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/b18dd2ce/attachment.html From fdelawarde at wirelessmundi.com Wed Jun 3 09:30:40 2009 From: fdelawarde at wirelessmundi.com (Francois Delawarde) Date: Wed, 03 Jun 2009 18:30:40 +0200 Subject: [Freeswitch-users] video transcoding Message-ID: <1244046640.28699.63.camel@localhost.localdomain> Hello, I'm interested in being able to do video transcoding mainly for bridging 3G mobile and sip networks, and maybe later on some conferencing with FS. Are video codecs planned to be added to FS even in a far future? Are there copyright/patent problems with common video codecs (H.263 / H.264) or with libraries (ffmpeg) that would prevent any of that from happening? Meanwhile, would it be feasible to do some video transcoding using external software (vlc?) with socket connections from-to FS? Thanks, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/c4d1d377/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 10:00:32 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:00:32 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> Message-ID: <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> When I type that verbatim, I get the following error: -ERR Unknown command! M On Wed, Jun 3, 2009 at 8:47 AM, Brian West wrote: > Thats not the problem.. > Phone registers to freeswitch has 192.168.0.100 in the sip packet... FS > sends a challenge back to 192.168.0.100 which obviously will FAIL cuz that > network isn't anywhere near the VPS box... > > > Go to the FreeSWITCH box and type "sofia profile internal siptrace on" and > I'll suspect you see this exact behavior. > > /b > > > > On Jun 3, 2009, at 10:42 AM, Matthew Lockwood wrote: > > SPI disabled. There's no obvious option to disable SIP ALG, but NAT > filtering was changed from secure to open. Problem still persists. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0ec49beb/attachment.html From brian at freeswitch.org Wed Jun 3 10:04:33 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 12:04:33 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> Message-ID: <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> Update to SVN trunk. /b On Jun 3, 2009, at 12:00 PM, Matthew Lockwood wrote: > When I type that verbatim, I get the following error: > > -ERR Unknown command! > > M Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/4dd8ad84/attachment-0001.html From matthew.lockwood at gmail.com Wed Jun 3 10:11:42 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:11:42 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> Message-ID: <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> Is that stable enough to use in production? On Wed, Jun 3, 2009 at 10:04 AM, Brian West wrote: > Update to SVN trunk. > /b > > On Jun 3, 2009, at 12:00 PM, Matthew Lockwood wrote: > > When I type that verbatim, I get the following error: > > -ERR Unknown command! > > M > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/9b57c7b9/attachment.html From intralanman at freeswitch.org Wed Jun 3 10:16:05 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 13:16:05 -0400 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> Message-ID: <4A26AFD5.1080903@freeswitch.org> Matthew Lockwood wrote: > Is that stable enough to use in production? it's more stable than "not working" -Ray From brian at freeswitch.org Wed Jun 3 10:34:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 12:34:12 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <4A26AFD5.1080903@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> <4A26AFD5.1080903@freeswitch.org> Message-ID: <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> It usually is... if you pop on IRC people can tell you one way or another. Issues never hang around for very long! /b On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote: > Matthew Lockwood wrote: >> Is that stable enough to use in production? > it's more stable than "not working" > -Ray Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/c599f0dc/attachment.html From matthew.lockwood at gmail.com Wed Jun 3 10:39:02 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:39:02 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> <4A26AFD5.1080903@freeswitch.org> <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> Message-ID: <415541b10906031039s49832117m8972e557f6875760@mail.gmail.com> It's installing now. I'll get back with the results shortly. On Wed, Jun 3, 2009 at 10:34 AM, Brian West wrote: > It usually is... if you pop on IRC people can tell you one way or another. > Issues never hang around for very long! > /b > > On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote: > > Matthew Lockwood wrote: > > Is that stable enough to use in production? > > it's more stable than "not working" > -Ray > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0d3cea27/attachment.html From testeador01 at gmail.com Wed Jun 3 10:39:25 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 3 Jun 2009 12:39:25 -0500 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: <20090603063909.GA19487@jdc.jasonjgw.net> References: <20090603063909.GA19487@jdc.jasonjgw.net> Message-ID: Hi Woody :) You cannot use the variable on another extension, however you could just merge both extensions' conditions. then your only problem would be that you're not exporting the value, after set, you gotta export, look at this example (a little extract from dialplan/default.xml): ... 2009/6/3 Jason White > Woody Dickson wrote: > > I am getting a strange problem in my dialplan. > > > > After doing "SET", I want to use it in the next condition field. But > then > > the value is not being set properly. > > When parsing the dial plan, FreeSWITCH tests all of the conditions, then > builds a linked list of actions to execute. Once this is done, the actions > are > executed, in order. > > This is why you can't simply set a variable in one extension and test it in > the condition of a later extension. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/64af6b64/attachment.html From mrene_lists at avgs.ca Wed Jun 3 10:43:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 13:43:18 -0400 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: <20090603063909.GA19487@jdc.jasonjgw.net> Message-ID: <2B642D7C-2779-4BF4-B3AF-017CDD4CE3CE@avgs.ca> Export transfers the variable to the B-leg whenever the channel is bridged, it doesnt affect how the dialplan work, conditions are still checked before executing anything. Math On 3-Jun-09, at 1:39 PM, Milena wrote: > Hi Woody :) > You cannot use the variable on another extension, however you could > just merge both extensions' conditions. > > then your only problem would be that you're not exporting the value, > after set, you gotta export, look at this example (a little extract > from dialplan/default.xml): > > > > > > > > > > > > ... > > > > > > > 2009/6/3 Jason White > Woody Dickson wrote: > > I am getting a strange problem in my dialplan. > > > > After doing "SET", I want to use it in the next condition field. > But then > > the value is not being set properly. > > When parsing the dial plan, FreeSWITCH tests all of the conditions, > then > builds a linked list of actions to execute. Once this is done, the > actions are > executed, in order. > > This is why you can't simply set a variable in one extension and > test it in > the condition of a later extension. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/49c442ef/attachment-0001.html From larclap at yahoo.com Wed Jun 3 11:04:22 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 3 Jun 2009 11:04:22 -0700 Subject: [Freeswitch-users] Dialplan appliction db? Message-ID: <053901c9e475$b91e3140$2b5a93c0$@com> Anyone point me to the wiki which describes the "db" application and its arguments? The following is a snippet from a dialplan. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/dc902e77/attachment.html From mrene_lists at avgs.ca Wed Jun 3 11:06:03 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 14:06:03 -0400 Subject: [Freeswitch-users] Dialplan appliction db? In-Reply-To: <053901c9e475$b91e3140$2b5a93c0$@com> References: <053901c9e475$b91e3140$2b5a93c0$@com> Message-ID: <327DDDBA-A6B8-4243-AAFE-260B466870E7@avgs.ca> freeswitch at Maths-Mac.local> show api db shAPI CALL [show(api db)] output: name,description,syntax,key db,db get/set,[insert|delete|select]///,mod_limit 1 total. freeswitch at Maths-Mac.local> show api hash API CALL [show(api hash)] output: name,description,syntax,key hash,hash get/set,[insert|delete|select]///,mod_limit 1 total. freeswitch at Maths-Mac.local> On 3-Jun-09, at 2:04 PM, Lars Zeb wrote: > Anyone point me to the wiki which describes the ?db? application and > its arguments? The following is a snippet from a dialplan. > > > > > > > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/cd42a759/attachment.html From larclap at yahoo.com Wed Jun 3 11:26:57 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 3 Jun 2009 11:26:57 -0700 Subject: [Freeswitch-users] Error reloadxml at console Message-ID: <055c01c9e478$e1163ff0$a342bfd0$@com> When I type reloadxml at fs console, I get the following message: freeswitch at fs> reloadxml API CALL [reloadxml()] output: +OK [[error near line 3182]: unclosed ). You warned me about this in an earlier email. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 03, 2009 11:52 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error reloadxml at console On Wed, Jun 3, 2009 at 11:26 AM, Lars Zeb wrote: When I type reloadxml at fs console, I get the following message: freeswitch at fs> reloadxml API CALL [reloadxml()] output: +OK [[error near line 3182]: unclosed when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/7b38836d/attachment.html From mike at jerris.com Thu Jun 4 01:59:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 04:59:34 -0400 Subject: [Freeswitch-users] Error causing freeswitch to crash In-Reply-To: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> References: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please attempt to reproduce this issue with trunk with crash protection disabled, and if you are able please file a jira with a backtrace of the crash Mike On Jun 4, 2009, at 4:23 AM, Andy Ayers wrote: > Hi, > > Every few days I'm getting this error which is causing Freeswitch to > crash. Can anyone tell me what may be causing this or how to prevent > it? > > 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 > handle_fatality() Caught signal 11 for unmapped thread! > > Many thanks > Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/6c2535aa/attachment.html From mike at jerris.com Thu Jun 4 02:02:52 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 05:02:52 -0400 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604082921.GA838@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> Message-ID: Can you please re-test with current svn trunk. we added some new nat busting code yesterday that may assist with this. You will need to specify the new param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 ) Mike On Jun 4, 2009, at 4:29 AM, Jason White wrote: > I hate NAT with a passion that strengthens by the day! > > I'm trying to interact with my ISP, which is a SIP provider. My FS > system is > behind a Cisco router (IOS 12.4(6)t). The provider recommends > turning off SIP > handling in the router's NAT configuration due to bugs in this > version of IOS. > I have found that I need to do this, otherwise incoming calls are > never > received, even though outgoing calls work without this change in > place. > > In the router: > no ip nat service sip udp port 5060 > ip nat inside source static tcp 192.168.0.2 5080 interface Dialer1 > 5080 > ip nat inside source static udp 192.168.0.2 5080 interface Dialer1 > 5080 > > With this change, incoming calls to FS are fine, but outgoing calls > are not > (see the SIP trace below). > > However, if I register to the same provider from an internal Snom > 320 phone > and make a call, it works. > > FreeSWITCH has sophisticated NAT handling features, as does the > other side of > this connection, and somehow they aren't working together. (I don't > know how > the other end is set up, but they claim to have complex NAT handling > logic). > > Registration is successful, by the way. > > The 118.208.xxx.xxx address is mine, dynamically allocated by the > ISP. The > xxxxxxxxxx at sip.internode.on.net address is my user name/address at > the service > provider. > > freeswitch at default> sofia profile external siptrace on > Enabled sip debugging on external > freeswitch at default> send 948 bytes to udp/[203.2.134.1]:5060 at > 07:14:51.882839: > > ------------------------------------------------------------------------ > REGISTER sip:sip.internode.on.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK4m9SgBce1DXpc > Max-Forwards: 70 > From: > ;tag=jy38KK47549va > To: > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931044 REGISTER > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj48x4rTpg7mg2BW", > cnonce="8SrBsct3EiylvAAaS8o90w", algorithm=MD5, uri="sip:sip.internode.on.net;transport=udp > ", response="85ec38442fbd26153bacd99c659bd037", qop=auth, nc=0000001c > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[203.2.134.1]:5060 at 07:14:51.967127: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK4m9SgBce1DXpc;rport=5080 > From: > ;tag=jy38KK47549va > To: > < > sip:xxxxxxxxxx > @sip.internode.on.net;transport=udp>;tag=aprqcauh8h3-4d3bh0p08vt9a > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931044 REGISTER > Contact: >;expires=60 > > > ------------------------------------------------------------------------ > 2009-06-04 17:14:53 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/internal/ > 1000 at 192.168.0.2:5070 [6653f750-50d7-11de-b1c2-25f4151d7bef] > 2009-06-04 17:14:53 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing Extension 1000->90871271202 in context default > 2009-06-04 17:14:53 [NOTICE] mod_sofia.c:1430 > sofia_receive_message() Ring-Ready sofia/internal/ > 1000 at 192.168.0.2:5070! > 2009-06-04 17:14:53 [NOTICE] mod_dptools.c:415 ring_ready_function() > Ring Ready sofia/internal/1000 at 192.168.0.2:5070! > 2009-06-04 17:14:53 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/external/0871271202 > [66551aea-50d7-11de-b1c2-25f4151d7bef] > send 1242 bytes to udp/[203.2.134.1]:5060 at 07:14:53.275254: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK5X2jj6vHypK9Q > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > recv 326 bytes from udp/[203.2.134.1]:5060 at 07:14:53.362415: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK5X2jj6vHypK9Q;rport=5080 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > > > ------------------------------------------------------------------------ > recv 487 bytes from udp/[203.2.134.1]:5060 at 07:14:53.371970: > > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK5X2jj6vHypK9Q;rport=5080 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: 0871271202 at sip.internode.on.net>;tag=1512759423-1244099693344 > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > WWW-Authenticate: DIGEST > qop > = > "auth > ",nonce > ="BroadWorksXfvj4u334T7pbfc4BW",algorithm=MD5,realm="BroadWorks" > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 388 bytes to udp/[203.2.134.1]:5060 at 07:14:53.372231: > > ------------------------------------------------------------------------ > ACK sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK5X2jj6vHypK9Q > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: 0871271202 at sip.internode.on.net>;tag=1512759423-1244099693344 > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 ACK > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:53.372622: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:53.873867: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:54.873857: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:56.873871: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:00.873875: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:08.879056: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:24.881893: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:25.375565: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:25.877859: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:26.877862: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:28.878002: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:32.881880: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 948 bytes to udp/[203.2.134.1]:5060 at 07:15:32.898046: > > ------------------------------------------------------------------------ > REGISTER sip:sip.internode.on.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK7FN4Nvyrr8ZeF > Max-Forwards: 70 > From: > ;tag=jy38KK47549va > To: > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931045 REGISTER > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj48x4rTpg7mg2BW", > cnonce="8SrBsct3EiylvAAaS8o90w", algorithm=MD5, uri="sip:sip.internode.on.net;transport=udp > ", response="2e9f91b4c82114b7c52a6c162633fbf2", qop=auth, nc=0000001d > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[203.2.134.1]:5060 at 07:15:33.017609: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK7FN4Nvyrr8ZeF;rport=5080 > From: > ;tag=jy38KK47549va > To: > < > sip:xxxxxxxxxx > @sip.internode.on.net;transport=udp>;tag=aprqcauh8h3-4d3bh0p08vtba > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931045 REGISTER > Contact: >;expires=60 > > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:40.882383: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > 2009-06-04 17:15:53 [NOTICE] switch_ivr_originate.c:1957 > switch_ivr_originate() Hangup sofia/external/0871271202 > [CS_CONSUME_MEDIA] [NO_ANSWER] > 2009-06-04 17:15:53 [INFO] mod_dptools.c:2091 > audio_bridge_function() Originate Failed. Cause: NO_ANSWER > 2009-06-04 17:15:53 [NOTICE] switch_core_state_machine.c:179 > switch_core_standard_on_execute() Hangup sofia/internal/1000 at 192.168.0.2 > :5070 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/internal/1000 at 192.168.0.2 > :5070) Ended > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/internal/1000 at 192.168.0.2 > :5070 [CS_DESTROY] > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 8 (sofia/external/0871271202) > Ended > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/external/0871271202 > [CS_DESTROY] > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:56.882385: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > > freeswitch at default> /exit > > jason at jdc:~$ exit > > Script done on Thu 04 Jun 2009 17:16:06 EST > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Jun 4 02:05:43 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 05:05:43 -0400 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <200906041628066407689@gmail.com> References: <200906041628066407689@gmail.com> Message-ID: <84AF0BA9-7027-4D51-B5FE-9672F67D71A2@jerris.com> Are you having this issue on your analog or pri lines? what does your openzap.conf look like? Mike On Jun 4, 2009, at 4:28 AM, god.nirvana wrote: > hi all > i am new to freeswitch. > there are some busy tone detect issues,i hope someone could help me. > i installed freeswitch from trunk,openzap,zaptel.... > but i found some busy tone isuues > > my tones.conf: > [us] > generate-dial => v=-7;%(1000,0,350,440) > detect-dial => 350,440 > generate-ring => v=-7;%(2000,4000,440,480) > detect-ring => 440,480 > generate-busy => v=-7;%(500,500,450,340) > detect-busy =>450,340 > generate-attn => v=0;%(100,100,1400,2060,2450,2600) > detect-attn => 1400,2060,2450,2600 > generate-callwaiting-sas => v=0;%(300,0,440) > detect-callwaiting-sas => 440 > generate-callwaiting-cas => v=0;%(80,0,2750,2130) > detect-callwaiting-cas => 2750,2130 > detect-fail1 => 913.8 > detect-fail2 => 1370.6 > detect-fail3 => 776.7 > > > openzap.conf.xml : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > when i call the pstn phone from a ip phone,if the pstn call hangup > first,the ip phone will hear the busy tone,but the system does not > handle the busytone ,the channel does not erase. so i have to add > in the dialplan.and it works.the channel > erased. > > but in the conference case,pstn phone call in,hangup. all > participants hear the tone,"do ~,do~~".freeswitch doest handle it. > so i change the conference dialplan. > > > > > > > > > > restart freeswitch,try again,freeswitch not handle the hangup tone > still,all participants hear the tone. > how to solve it?could some one help me ??? > thx! > BR > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/72bef1c1/attachment.html From dujinfang at gmail.com Thu Jun 4 02:13:01 2009 From: dujinfang at gmail.com (seven) Date: Thu, 4 Jun 2009 17:13:01 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: Message-ID: the default config allows 1002 press *1 and 1003 to do blind transfer, also you may interest the att_xfer, see dp_tools on wiki. On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: > If i don't want to use softphone function to transfer the call ,how > to do it?? > > 2009/6/4 Brian West > Depends.. Press the transfer key on your phone is how I would do > it.. what kind of phone do you have? > > /b > > On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: > >> When User(1001) calling with User(1002) , >> >> how to transfer User(1002) to User(1003)?? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/772a0d3d/attachment.html From god.nirvana at gmail.com Thu Jun 4 02:15:12 2009 From: god.nirvana at gmail.com (god.nirvana) Date: Thu, 4 Jun 2009 17:15:12 +0800 Subject: [Freeswitch-users] busy tone detect issue References: <200906041628066407689@gmail.com>, <84AF0BA9-7027-4D51-B5FE-9672F67D71A2@jerris.com> Message-ID: <200906041715100317684@gmail.com> hi,thanks for your reply. my openzap.conf like this: [span zt FXO1] name => OpenZAP-FXO1 number => 1 fxo-channel => 1 [span zt FXO2] name => OpenZAP-FXO2 number => 2 fxo-channel => 2 [span zt FXO2] name => OpenZAP-FXO2 number => 3 fxo-channel => 3 [span zt FXO2] name => OpenZAP-FXO2 number => 3 fxo-channel => 4 i have a 4 fxo ports TDM400. 2009-06-04 god.nirvana ???? Michael Jerris ????? 2009-06-04 17:06:05 ???? freeswitch-users ??? ??? Re: [Freeswitch-users] busy tone detect issue Are you having this issue on your analog or pri lines? what does your openzap.conf look like? Mike On Jun 4, 2009, at 4:28 AM, god.nirvana wrote: hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 openzap.conf.xml : when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b0ad3d1f/attachment-0001.html From brad.tuan at gmail.com Thu Jun 4 02:27:10 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Thu, 4 Jun 2009 17:27:10 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> Message-ID: I know why the display name is wrong.......... in conf\directory\97730002.xml I forgot this setting........... but if I don't set cidr here ,the call from 163.28.32.51 can't come into my FS. How to make some setting for this?? 2009/6/3 Anthony Minessale > also press f8 before you take the console log to get the debugging info > and paste the resulting trace in http://pastebin.freeswitch.org rather > than right in the email > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/5c0f8fe0/attachment.html From brad.tuan at gmail.com Thu Jun 4 02:36:02 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Thu, 4 Jun 2009 17:36:02 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: Message-ID: I mean when 1001 and 1002 are talking to each other , then 1001 want to transfer 1002 to 1003. 2009/6/4 seven > the default config allows 1002 press *1 and 1003 to do blind transfer, also > you may interest the att_xfer, see dp_tools on wiki. > > > On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: > > If i don't want to use softphone function to transfer the call ,how to do > it?? > > 2009/6/4 Brian West > >> Depends.. Press the transfer key on your phone is how I would do it.. what >> kind of phone do you have? >> /b >> >> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >> >> When User(1001) calling with User(1002) , >> >> how to transfer User(1002) to User(1003)?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/12f07146/attachment.html From jason at jasonjgw.net Thu Jun 4 02:41:52 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 4 Jun 2009 19:41:52 +1000 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: References: <20090604082921.GA838@jdc.jasonjgw.net> Message-ID: <20090604094152.GA15717@jdc.jasonjgw.net> Michael Jerris wrote: > Can you please re-test with current svn trunk. we added some new nat > busting code yesterday that may assist with this. You will need to > specify the new > param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 > ) If I apply this to the external profile (using it on the internal profile has no effect), the Via headers now receive the external IP address, i.e., the publicly routable address provided by the ISP. However, the session description still has the private address in it, which causes the remote end to issue the following: SIP/2.0 488 Invalid Session Description Warning: 301 203.2.134.1 "invalid transport IP address" I assume (but may be wrong - full traces can be provided if necessary) that the problem is in the SDP that we're sending out: o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 Can this be fixed up? Thanks. Jason. From zhaoxxqq at 163.com Thu Jun 4 02:44:34 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Thu, 4 Jun 2009 17:44:34 +0800 Subject: [Freeswitch-users] (no subject) Message-ID: <200906041744323372036@163.com> zhaoxxqq pisze: > HI, > I use event socket to send command to FS conference. > I send " conference testconf play /root/test.wav" in console. It worked ok. > I send "api conference testconf play /record/test.wav" by event socket. > and the response is"Disconneted, Good bye.See you at ClueCon..". > I changed the wav file to www root. the same problem. can you help me? > 2009-06-01 Do you use 'auth ClueCon' before sending 'api' command? Szymon I have not use 'auth Cluecon' before sending api command. I send other api have no problem.only play wav have problems 2009-06-04 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/05377f5b/attachment.html From jason at jasonjgw.net Thu Jun 4 03:39:58 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 4 Jun 2009 20:39:58 +1000 Subject: [Freeswitch-users] api conference play command In-Reply-To: <200906041744323372036@163.com> References: <200906041744323372036@163.com> Message-ID: <20090604103958.GA25407@jdc.jasonjgw.net> zhaoxxqq wrote: > I have not use 'auth Cluecon' before sending api command. > I send other api have no problem.only play wav have problems Try it from a telnet session. Start with auth ClueCon, then issue the API command as shown in my example. Unless you do something wrong, it will work. From dujinfang at gmail.com Thu Jun 4 03:51:21 2009 From: dujinfang at gmail.com (dujinfang) Date: Thu, 4 Jun 2009 18:51:21 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: Message-ID: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> yes. Did you ever tried that? On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: > I mean when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003. > 2009/6/4 seven > the default config allows 1002 press *1 and 1003 to do blind > transfer, also you may interest the att_xfer, see dp_tools on wiki. > > > On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >> If i don't want to use softphone function to transfer the call ,how >> to do it?? >> >> 2009/6/4 Brian West >> Depends.. Press the transfer key on your phone is how I would do >> it.. what kind of phone do you have? >> >> /b >> >> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >> >>> When User(1001) calling with User(1002) , >>> >>> how to transfer User(1002) to User(1003)?? >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3cf2787f/attachment-0001.html From yivzhenko at mksat.net Thu Jun 4 04:02:42 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 4 Jun 2009 14:02:42 +0300 Subject: [Freeswitch-users] mod_lcr caller id Message-ID: <200906041402.43704.yivzhenko@mksat.net> Hello. I have another problem with setting caller id with cid field. command lcr has parameter [caller_id]. and if i call lcr from the commandline it works fine, but if i call lcr from dialplan it ignores this parameter and use original caller id from caller. my cid field = /^(.*)$/999$1/ If i call lcr from the commandline lcr 444555 default 11111 lcr return origination_caller_id_number=99911111 but if i call lcr from dialplan lcr return origination_caller_id_number=9991002 1002 is a original caller id number There is a way to use this parameter from dialplan? Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/76765413/attachment.html From jim at evolutiontel.net Thu Jun 4 05:43:24 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 22:43:24 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. Message-ID: Hi All, Looking for some debugging tips and comments on what might be causing the media port in the 200OK ( Answer message) to be set to 0 by freeswitch. Essentially it looks like data might be getting trampled somehow. Portion of 200OK going into Freeswitch m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 NSE/8000..a= fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:20..a=sendrecv.. Portion of 200OK coming out of Freeswitch m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-15.. Note the media port has been set to 0 and the rtpmap for G729 is also not correct. On receipt of this bad 200Ok the originator sends a BYE. We are using FS as a B2BUA with bypass_media set to true. Thus IMHO Freeswitch should not be touching the SDP portion of the message and just passing it through. This can reproduce this at will, so I can collect whatever data is nessicary. I have added the sofia debug from the console. Thanks, -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net -------------- next part -------------- tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9bcba58 from (udp/192.168.0.2:5070) has 1303 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9bcba58 (1303 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received INVITE sip:1234567891 at 192.168.0.2:5060 SIP/2.0 (CSeq 811801) nta: INVITE (811801) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9bfa748) called soa_set_params(static::0x9bcca70, ...) called nta_leg_tcreate(0x9c19f60) soa_init_offer_answer(static::0x9bcca70) called soa_set_remote_sdp(static::0x9bcca70, (nil), 0x9ce1541, 286) called nua(0x9bfa748): adding session usage tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 705 bytes of 705 to udp/192.168.0.2:5060 tport_vsend returned 705 nta: sent 100 Trying for INVITE (811801) nua(0x9bfa748): event i_invite 100 Trying nua(0x9bfa748): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x9bcca70, [0xb77b7bac], [0xb77b7ba8], [(nil)]) called nua(0x9bfa748): event i_state 100 Trying nua: nua_application_event: entering 2009-06-04 21:58:34 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/987654321 at sip.evolutiontel.net [6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b] nua: nua_handle_bind: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [received][100] 2009-06-04 21:58:34 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 570753038 570753038 IN IP4 202.xx.xxx.xx s=ENSResip c=IN IP4 202.xx.xxx.xx t=0 0 m=audio 13298 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - 2009-06-04 21:58:34 [DEBUG] sofia.c:3182 sofia_handle_sip_i_state() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_NEW -> CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State INIT 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/987654321 at sip.evolutiontel.net SOFIA INIT 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_INIT -> CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State INIT going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State ROUTING 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/987654321 at sip.evolutiontel.net SOFIA ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/987654321 at sip.evolutiontel.net Standard ROUTING 2009-06-04 21:58:34 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 987654321->1234567891 in context evolutiontel Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unloop] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->global] continue=true Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${network_addr}(192.168.0.2) =~ /^$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net ANTI-Action set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [global] ${numbering_plan}() =~ /^$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set_user(default@${domain_name}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_h_X-ZRTP-On}() =~ /^Y$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_secure_media}() =~ /^true$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_user_agent}(ENSR2.5.4) =~ /^PolycomSound(Point|Station)IP-S(S|P)IP_\d{3,4}-UA\/((3).(\d).(\d).(\d{4}))$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Absolute Condition [global] Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->vmain] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [vmain] destination_number(1234567891) =~ /^121/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->vmain1] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [vmain1] destination_number(1234567891) =~ /^123/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->vmain2] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [vmain2] destination_number(1234567891) =~ /^122/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->ivr_demo] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [ivr_demo] destination_number(1234567891) =~ /^5000$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(1234567891) =~ /^5900$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(1234567891) =~ /^5901$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(1234567891) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(1234567891) =~ /^parking$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(1234567891) =~ /callpark/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(1234567891) =~ /pickup/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->wait] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [wait] destination_number(1234567891) =~ /^wait$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->National_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [National_calls] destination_number(1234567891) =~ /^0(2|3|4|5|7|8|9)[0-9]{8}$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->Special_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [Special_calls] destination_number(1234567891) =~ /^1[3|8][0-9]+$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->International_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [International_calls] destination_number(1234567891) =~ /^0011[0-9]+$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->On-Net_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [On-Net_calls] destination_number(1234567891) =~ /^063[0-9]{7}$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(execute_on_answer=sched_hangup +${sip_h_x-max-timer} ALLOTED_TIMEOUT) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(sip_cid_type=pid) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(continue_on_fail=79) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(bypass_media=false) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(ringback=%(400,200,401,450);%(400,2200,400,450)) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action pre_answer() Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action export(sip_secure_media=true) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_ROUTING -> CS_EXECUTE 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State ROUTING going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_EXECUTE 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State EXECUTE 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/987654321 at sip.evolutiontel.net SOFIA EXECUTE 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/987654321 at sip.evolutiontel.net Standard EXECUTE EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(use_profile=nat) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [use_profile]=[nat] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set_user(default at 192.168.0.2) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(bypass_media=true) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(bypass_media=true) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net hash(insert/192.168.0.2-spymap/987654321/6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/987654321/1234567891) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/global/6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(execute_on_answer=sched_hangup +7200 ALLOTED_TIMEOUT) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [execute_on_answer]=[sched_hangup +7200 ALLOTED_TIMEOUT] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(sip_cid_type=pid) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [sip_cid_type]=[pid] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(continue_on_fail=79) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [continue_on_fail]=[79] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net bridge({sip_from_uri=sip:987654321 at sip.evolutiontel.net}sofia/internal/1234567891 at 192.168.0.2^0312341234 at 202.xxx.xx.xx:5060) 2009-06-04 21:58:34 [DEBUG] switch_ivr_originate.c:1017 switch_ivr_originate() variable string 0 = [sip_from_uri=sip:987654321 at sip.evolutiontel.net] 2009-06-04 21:58:34 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/1234567891 at 192.168.0.2 [64f91098-4f97-4c17-b95b-0d1693c64f8a] 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:2719 sofia_outgoing_channel() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State INIT 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1234567891 at 192.168.0.2 SOFIA INIT nua: nh_create_handle: entering nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x9d17500): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9d17500) called soa_set_params(static::0x9bf1e20, ...) called soa_set_params(static::0x9bf1e20, ...) called soa_set_user_sdp(static::0x9bf1e20, (nil), 0x9c1d48f, -1) called soa_set_capability_sdp(static::0x9bf1e20, (nil), 0x9c1d48f, -1) called nua(0x9d17500): adding session usage nta_leg_tcreate(0x9bce5c8) soa_init_offer_answer(static::0x9bf1e20) called soa_generate_offer(static::0x9bf1e20, 0) called soa_static_offer_answer_action(0x9bf1e20, soa_generate_offer): called soa_static(0x9bf1e20, soa_generate_offer): generating local description soa_static(0x9bf1e20, soa_generate_offer): upgrade with local description soa_sdp_mode_set(0xb77b7ef4, (nil), ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.125.42.98 (a local address) soa_static(0x9bf1e20, soa_generate_offer): storing local description soa_get_local_sdp(static::0x9bf1e20, [(nil)], [0xb77b7f7c], [0xb77b7f78]) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 1233 bytes of 1233 to udp/192.168.0.2:5060 tport_vsend returned 1233 nta: sent INVITE (115940021) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9cd15d8 for udp/192.168.0.2:5070 (already 0) nua(0x9d17500): call state changed: init -> calling, sent offer soa_get_local_sdp(static::0x9bf1e20, [0xb77b7fa4], [0xb77b7fa0], [(nil)]) called nua(0x9d17500): event i_state INVITE sent nua: nua_application_event: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9d5c900 from (udp/192.168.0.2:5070) has 303 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9d5c900 (303 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 100 Trying for INVITE (115940021) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 1.918 ms tport_release(0x9bb0e28): 0x9cd15d8 by 0x9d5c840 with 0x9d5c900 (preliminary) nua(0x9d17500): sent signal r_invite 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/1234567891 at 192.168.0.2 entering state [calling][0] nua: nua_handle_magic: entering 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State INIT going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State ROUTING 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1234567891 at 192.168.0.2 SOFIA ROUTING 2009-06-04 21:58:34 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State ROUTING going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_CONSUME_MEDIA 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State CONSUME_MEDIA 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State CONSUME_MEDIA going to sleep tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c84f30 from (udp/192.168.0.2:5070) has 711 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9c84f30 (711 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 180 Ringing for INVITE (115940021) nta: 180 Ringing is going to a transaction tport_release(0x9bb0e28): 0x9cd15d8 by 0x9d5c840 with 0x9c84f30 (preliminary) nua(0x9d17500): event r_invite 180 Ringing nua(0x9d17500): call state changed: calling -> proceeding nua(0x9d17500): event i_state 180 Ringing nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/1234567891 at 192.168.0.2 entering state [proceeding][180] 2009-06-04 21:58:34 [NOTICE] sofia.c:3103 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1234567891 at 192.168.0.2! 2009-06-04 21:58:34 [DEBUG] sofia.c:3111 sofia_handle_sip_i_state() sofia/internal/987654321 at sip.evolutiontel.net receive message [RINGING] nua: nua_respond: entering nua(0x9bfa748): sent signal r_respond 2009-06-04 21:58:34 [NOTICE] mod_sofia.c:1422 sofia_receive_message() Ring-Ready sofia/internal/987654321 at sip.evolutiontel.net! 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering nua(0x9bfa748): recv signal r_respond 180 Ringing nua: nua_stack_set_params: entering soa_set_params(static::0x9bcca70, ...) called nua: nua_invite_server_respond: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1132 bytes of 1132 to udp/192.168.0.2:5060 tport_vsend returned 1132 nta: sent 180 Ringing for INVITE (811801) nua(0x9bfa748): call state changed: received -> early nua(0x9bfa748): event i_state 180 Ringing nua: nua_application_event: entering 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [early][180] nua: nua_handle_magic: entering 2009-06-04 21:58:34 [DEBUG] switch_ivr_originate.c:1768 switch_ivr_originate() sofia/internal/987654321 at sip.evolutiontel.net receive message [RINGING] 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:34 [NOTICE] switch_ivr_originate.c:1768 switch_ivr_originate() Ring Ready sofia/internal/987654321 at sip.evolutiontel.net! nta: timer set next to 59934 ms tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c855e8 from (udp/192.168.0.2:5070) has 1215 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9c855e8 (1215 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for INVITE (115940021) nta: 200 OK is going to a transaction tport_release(0x9bb0e28): 0x9cd15d8 by 0x9d5c840 with 0x9c855e8 nta: timer shortened to 32000 ms soa_set_remote_sdp(static::0x9bf1e20, (nil), 0x9d149c5, 306) called soa_process_answer(static::0x9bf1e20) called soa_static_offer_answer_action(0x9bf1e20, soa_process_answer): called soa_sdp_mode_set(0x9d197c0, 0x9cec6b0, ""): called soa_static(0x9bf1e20, soa_process_answer): upgrade codecs with remote description soa_static(0x9bf1e20, soa_process_answer): storing local description soa_activate(static::0x9bf1e20, (nil)) called nua(0x9d17500): INVITE: processed SDP answer in 200 OK nua(0x9d17500): event r_invite 200 OK nua: nua_application_event: entering nua: nua_handle_magic: entering soa_activate(static::0x9bf1e20, (nil)) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 621 bytes of 621 to udp/192.168.0.2:5060 tport_vsend returned 621 nta: sent ACK (115940021) to */192.168.0.2:5060 nua(0x9d17500): call state changed: proceeding -> ready, received answer soa_get_remote_sdp(static::0x9bf1e20, [0xb77b7bfc], [0xb77b7bf8], [(nil)]) called soa_get_params(static::0x9bf1e20, ...) called nua(0x9d17500): event i_state 200 OK nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/1234567891 at 192.168.0.2 entering state [ready][200] 2009-06-04 21:58:36 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 4495350 4495350 IN IP4 192.168.0.10 s=- c=IN IP4 60.241.91.137 t=0 0 m=audio 16580 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=direction:active a=oldmediaip:192.168.0.10 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1875 switch_channel_perform_mark_answered() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [NOTICE] sofia.c:3509 sofia_handle_sip_i_state() Channel [sofia/internal/1234567891 at 192.168.0.2] has been answered 2009-06-04 21:58:36 [DEBUG] sofia.c:3522 sofia_handle_sip_i_state() sofia/internal/987654321 at sip.evolutiontel.net receive message [ANSWER] nua: nua_respond: entering nua(0x9bfa748): sent signal r_respond 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [NOTICE] sofia.c:3522 sofia_handle_sip_i_state() Channel [sofia/internal/987654321 at sip.evolutiontel.net] has been answered 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1911 switch_channel_perform_mark_answered() sofia/internal/987654321 at sip.evolutiontel.net execute on answer: sched_hangup(+7200 ALLOTED_TIMEOUT) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net sched_hangup(+7200 ALLOTED_TIMEOUT) 2009-06-04 21:58:36 [DEBUG] switch_scheduler.c:214 switch_scheduler_add_task() Added task 6 switch_ivr_schedule_hangup (6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) to run at 1244123916 nua: nua_handle_magic: entering 2009-06-04 21:58:36 [DEBUG] switch_ivr_originate.c:2024 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/1234567891 at 192.168.0.2] 2009-06-04 21:58:36 [DEBUG] switch_ivr_bridge.c:791 switch_ivr_signal_bridge() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_ivr_bridge.c:792 switch_ivr_signal_bridge() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State EXECUTE going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HIBERNATE 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/987654321 at sip.evolutiontel.net SOFIA HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/987654321 at sip.evolutiontel.net Standard HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HIBERNATE going to sleep nua(0x9d17500): event i_active 200 Call active nua(0x9bfa748): recv signal r_respond 200 OK nua: nua_stack_set_params: entering soa_set_params(static::0x9bcca70, ...) called soa_set_user_sdp(static::0x9bcca70, (nil), 0x9d739a8, -1) called soa_set_capability_sdp(static::0x9bcca70, (nil), 0x9d739a8, -1) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x9bcca70) called soa_static_offer_answer_action(0x9bcca70, soa_generate_answer): called soa_static(0x9bcca70, soa_generate_answer): generating local description soa_static(0x9bcca70, soa_generate_answer): upgrade with remote description soa_static(0x9bcca70, soa_generate_answer): marking rejected media soa_sdp_mode_set(0xb77b7fb4, 0x9c2c578, ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.125.42.98 (a local address) soa_static(0x9bcca70, soa_generate_answer): storing local description soa_activate(static::0x9bcca70, (nil)) called soa_get_local_sdp(static::0x9bcca70, [(nil)], [0xb77b803c], [0xb77b8038]) called tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1421 bytes of 1421 to udp/192.168.0.2:5060 tport_vsend returned 1421 nta: sent 200 OK for INVITE (811801) nta: timer shortened to 500 ms nua(0x9bfa748): call state changed: early -> completed, sent answer soa_get_local_sdp(static::0x9bcca70, [0xb77b8124], [0xb77b8120], [(nil)]) called soa_get_params(static::0x9bcca70, ...) called nua(0x9bfa748): event i_state 200 OK 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HIBERNATE 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/1234567891 at 192.168.0.2 SOFIA HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/1234567891 at 192.168.0.2 Standard HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HIBERNATE going to sleep nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [completed][200] nua: nua_handle_magic: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9ce6898 from (udp/192.168.0.2:5070) has 479 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9ce6898 (479 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received ACK sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 811801) nta: ACK (811801) is going to INVITE (811801) nua: process_ack_or_cancel: entering soa_clear_remote_sdp(static::0x9bcca70) called nua(0x9bfa748): event i_ack 200 OK nua(0x9bfa748): call state changed: completed -> ready nua(0x9bfa748): event i_state 200 OK nua(0x9bfa748): event i_active 200 Call active nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [ready][200] nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9ce6898 from (udp/192.168.0.2:5070) has 479 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9ce6898 (479 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 811802) nta: canonizing sip:mod_sofia at 192.168.0.2:5070 with contact nta: BYE (811802) going to existing leg nua: nua_stack_process_request: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 593 bytes of 593 to udp/192.168.0.2:5060 tport_vsend returned 593 nta: sent 200 OK for BYE (811802) nua(0x9bfa748): event i_bye 200 Session Terminated nua(0x9bfa748): removing session usage nua(0x9bfa748): call state changed: ready -> terminated nua(0x9bfa748): event i_state 200 Session Terminated nua(0x9bfa748): event i_terminated 200 Session Terminated soa_destroy(static::0x9bcca70) called nta_leg_destroy(0x9c19f60) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [terminated][200] 2009-06-04 21:58:36 [NOTICE] sofia.c:3599 sofia_handle_sip_i_state() Hangup sofia/internal/987654321 at sip.evolutiontel.net [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/987654321 at sip.evolutiontel.net [KILL] 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua(0x9bfa748): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua: nua_application_event: entering nua(0x9bfa748): event i_terminated dropped 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_HANGUP 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HANGUP 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/987654321 at sip.evolutiontel.net Overriding SIP cause 480 with 200 from the other leg 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/987654321 at sip.evolutiontel.net hanging up, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [NOTICE] switch_ivr_bridge.c:712 signal_bridge_on_hangup() Hangup sofia/internal/1234567891 at 192.168.0.2 [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/1234567891 at 192.168.0.2 [KILL] 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/987654321 at sip.evolutiontel.net Standard HANGUP, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HANGUP going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_HANGUP -> CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/987654321 at sip.evolutiontel.net) State REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/987654321 at sip.evolutiontel.net Standard REPORTING, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/987654321 at sip.evolutiontel.net) State REPORTING going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_REPORTING -> CS_DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/internal/987654321 at sip.evolutiontel.net) Locked, Waiting on external entities 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/987654321 at sip.evolutiontel.net) Ended 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/987654321 at sip.evolutiontel.net [CS_DESTROY] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/987654321 at sip.evolutiontel.net) State DESTROY 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/987654321 at sip.evolutiontel.net SOFIA DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/987654321 at sip.evolutiontel.net Standard DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/987654321 at sip.evolutiontel.net) State DESTROY going to sleep nua(0x9bfa748): recv signal r_destroy nta_leg_destroy((nil)) 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_HANGUP 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HANGUP 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/1234567891 at 192.168.0.2 Overriding SIP cause 480 with 200 from the other leg 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/1234567891 at 192.168.0.2 hanging up, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:394 sofia_on_hangup() Sending BYE to sofia/internal/1234567891 at 192.168.0.2 nua: nua_bye: entering nua(0x9d17500): sent signal r_bye 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1234567891 at 192.168.0.2 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HANGUP going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1234567891 at 192.168.0.2) State REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/1234567891 at 192.168.0.2 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1234567891 at 192.168.0.2) State REPORTING going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 8 (sofia/internal/1234567891 at 192.168.0.2) Locked, Waiting on external entities 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 8 (sofia/internal/1234567891 at 192.168.0.2) Ended 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1234567891 at 192.168.0.2 [CS_DESTROY] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1234567891 at 192.168.0.2) State DESTROY 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/1234567891 at 192.168.0.2 SOFIA DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/1234567891 at 192.168.0.2 Standard DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1234567891 at 192.168.0.2) State DESTROY going to sleep nua(0x9d17500): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x9bf1e20, ...) called soa_terminate(static::0x9bf1e20) called soa_init_offer_answer(static::0x9bf1e20) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 871 bytes of 871 to udp/192.168.0.2:5060 tport_vsend returned 871 nta: sent BYE (115940022) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9d19bc8 for udp/192.168.0.2:5070 (already 0) 2009-06-04 21:58:36 [DEBUG] switch_scheduler.c:138 task_thread_loop() Deleting task 6 switch_ivr_schedule_hangup (6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c84f30 from (udp/192.168.0.2:5070) has 449 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9c84f30 (449 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for BYE (115940022) nta: 200 OK is going to a transaction nta_outgoing: RTT is 97.734 ms tport_release(0x9bb0e28): 0x9d19bc8 by 0x9bb6ef8 with 0x9c84f30 nua(0x9d17500): event r_bye 200 OK nua(0x9d17500): call state changed: terminating -> terminated nua(0x9d17500): event i_state 200 to BYE nua(0x9d17500): event i_terminated 200 to BYE nua(0x9d17500): removing session usage soa_destroy(static::0x9bf1e20) called nta_leg_destroy(0x9bce5c8) nua: terminated session 0x9d17500 nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x9d17500): sent signal r_destroy nua: nua_application_event: entering nua(0x9d17500): event i_terminated dropped nua(0x9d17500): recv signal r_destroy nta_leg_destroy((nil)) nta: timer set next to 4634 ms tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9ce6aa0 from (udp/192.168.0.2:5070) has 479 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9ce6aa0 (479 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 811802) nta: BYE (811802) going to existing BYE transaction nta: re-received BYE request, retransmitting 200 reply tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 593 bytes of 593 to udp/192.168.0.2:5060 tport_vsend returned 593 nta: timer I fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0xb77b81cc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 107 ms nta: timer K fired, terminate BYE (115940022) outgoing_reclaim_all((nil), (nil), 0xb77b81c8) nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/2 term, 1/3 free nta: timer set next to 26750 ms /exit [root at sip01 bin]# From anthony.minessale at gmail.com Thu Jun 4 05:49:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 07:49:41 -0500 Subject: [Freeswitch-users] Error causing freeswitch to crash In-Reply-To: References: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> Message-ID: <191c3a030906040549y2ec5fe8aica40122032057715@mail.gmail.com> also make sure you have a clean update to SVN trunk before re-testing. On Thu, Jun 4, 2009 at 3:59 AM, Michael Jerris wrote: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > Please attempt to reproduce this issue with trunk with crash protection > disabled, and if you are able please file a jira with a backtrace of the > crash > > Mike > > On Jun 4, 2009, at 4:23 AM, Andy Ayers wrote: > > Hi, > > Every few days I'm getting this error which is causing Freeswitch to crash. > Can anyone tell me what may be causing this or how to prevent it? > > 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 > handle_fatality() Caught signal 11 for unmapped thread! > > Many thanks > Andy > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/be334034/attachment.html From rupa at rupa.com Thu Jun 4 05:54:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 4 Jun 2009 07:54:24 -0500 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: <200906041402.43704.yivzhenko@mksat.net> References: <200906041402.43704.yivzhenko@mksat.net> Message-ID: The API mthod of setting the callerid is really for testing purposes. Since you have no callerid from the FS command line interface I added a parameter you can pass to specify the CID. When in the dialplan you actually have the CID, so none is used. mod_lcr is just grabbing the callerid on the channel. I assme this is not the base callerid you want passed to the sip provider? On Thu, Jun 4, 2009 at 6:02 AM, Yuriy Ivzhenko wrote: > Hello. > > I have another problem with setting caller id with cid field. > > command lcr has parameter [caller_id]. > > and if i call lcr from the commandline it works fine, > > but if i call lcr from dialplan it ignores this parameter and use original > caller id from caller. > > my cid field = /^(.*)$/999$1/ > > If i call lcr from the commandline > > lcr 444555 default 11111 > > lcr return origination_caller_id_number=99911111 > > but if i call lcr from dialplan > > > > lcr return origination_caller_id_number=9991002 > > 1002 is a original caller id number > > There is a way to use this parameter from dialplan? > > Yuriy > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/2e1d8f0c/attachment.html From anthony.minessale at gmail.com Thu Jun 4 05:55:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 07:55:52 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: <191c3a030906040555k1168c03an4330235ab8b69320@mail.gmail.com> Did you try what he said? The new nat busting stuff will allow you to use just the internal profile for everything. you may need to join irc and ask bkw for the proper config options. On Thu, Jun 4, 2009 at 4:41 AM, Jason White wrote: > Michael Jerris wrote: > > Can you please re-test with current svn trunk. we added some new nat > > busting code yesterday that may assist with this. You will need to > > specify the new > > param in the sofia profile (see > http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 > > ) > > If I apply this to the external profile (using it on the internal profile > has > no effect), the Via headers now receive the external IP address, i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d880d079/attachment.html From jim at evolutiontel.net Thu Jun 4 05:59:40 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 22:59:40 +1000 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: This is the one you need to change c=IN IP4 192.168.0.2. o= is just the owner whereas c= is the connection IP. Try rtp-ip in your sip profile Having said that Warning: 301 203.2.134.1 "invalid transport IP address" isnt 203.2.134.1 the address it is complaining about? On Thu, Jun 4, 2009 at 7:41 PM, Jason White wrote: > Michael Jerris wrote: >> Can you please re-test with current svn trunk. ?we added some new nat >> busting code yesterday that may assist with this. ?You will need to >> specify the new > ?> param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 >> ? ) > > If I apply this to the external profile (using it on the internal profile has > no effect), the Via headers now receive the external IP address, i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From anthony.minessale at gmail.com Thu Jun 4 05:59:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 07:59:13 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: Message-ID: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> you should have also turned in the sip trace sofia profile internal siptrace on On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke wrote: > Hi All, > > Looking for some debugging tips and comments on what might be causing > the media port in the 200OK ( Answer message) to be set to 0 by > freeswitch. Essentially it looks like data might be getting trampled > somehow. > > Portion of 200OK going into Freeswitch > m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 > NSE/8000..a= > fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 > 0-15..a=ptime:20..a=sendrecv.. > > Portion of 200OK coming out of Freeswitch > m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 > NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event > /8000..a=fmtp:101 0-15.. > > Note the media port has been set to 0 and the rtpmap for G729 is also > not correct. On receipt of this bad 200Ok the originator sends a BYE. > > We are using FS as a B2BUA with bypass_media set to true. Thus IMHO > Freeswitch should not be touching the SDP portion of the message and > just passing it through. > > This can reproduce this at will, so I can collect whatever data is > nessicary. I have added the sofia debug from the console. > > Thanks, > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d8a4ddaa/attachment-0001.html From jim at evolutiontel.net Thu Jun 4 06:09:18 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 23:09:18 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> Message-ID: I have traces using ngrep, however if you want to see it all in one file I will collect tommorow. Regards, On Thu, Jun 4, 2009 at 10:59 PM, Anthony Minessale wrote: > you should have also turned in the sip trace > sofia profile internal siptrace on > > > On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke wrote: >> >> Hi All, >> >> Looking for some debugging tips and comments on what might be causing >> the media port in the 200OK ( Answer message) to be set to 0 by >> freeswitch. ?Essentially it looks like data might be getting trampled >> somehow. >> >> Portion of 200OK going into Freeswitch >> m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 >> NSE/8000..a= >> ?fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >> 0-15..a=ptime:20..a=sendrecv.. >> >> Portion of 200OK coming out of Freeswitch >> m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 >> NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event >> ?/8000..a=fmtp:101 0-15.. >> >> Note the media port has been set to 0 and the rtpmap for G729 is also >> not correct. ?On receipt of this bad 200Ok the originator sends a BYE. >> >> We are using FS as a B2BUA with bypass_media set to true. ?Thus IMHO >> Freeswitch should not be touching the SDP portion of the message and >> just passing it through. >> >> This can reproduce this at will, so I can collect whatever data is >> nessicary. ?I have added the sofia debug from the console. >> >> Thanks, >> -- >> Jim Burke >> Director Evolutiontel. >> http://www.evolutiontel.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From jim at evolutiontel.net Thu Jun 4 06:10:23 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 23:10:23 +1000 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: Apologies that should have been ext-rtp-ip Regards, On Thu, Jun 4, 2009 at 10:59 PM, Jim Burke wrote: > This is the one you need to change c=IN IP4 192.168.0.2. ?o= is just > the owner whereas c= is the connection IP. ?Try rtp-ip in your sip > profile > > Having said that Warning: 301 203.2.134.1 "invalid transport IP > address" isnt 203.2.134.1 the address it is complaining about? > > On Thu, Jun 4, 2009 at 7:41 PM, Jason White wrote: >> Michael Jerris wrote: >>> Can you please re-test with current svn trunk. ?we added some new nat >>> busting code yesterday that may assist with this. ?You will need to >>> specify the new >> ?> param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 >>> ? ) >> >> If I apply this to the external profile (using it on the internal profile has >> no effect), the Via headers now receive the external IP address, i.e., the >> publicly routable address provided by the ISP. >> >> However, the session description still has the private address in it, which >> causes the remote end to issue the following: >> SIP/2.0 488 Invalid Session Description >> Warning: 301 203.2.134.1 "invalid transport IP address" >> >> I assume (but may be wrong - full traces can be provided if necessary) that >> the problem is in the SDP that we're sending out: >> o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 >> >> Can this be fixed up? >> >> Thanks. >> >> Jason. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From yivzhenko at mksat.net Thu Jun 4 06:15:15 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 4 Jun 2009 16:15:15 +0300 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: References: <200906041402.43704.yivzhenko@mksat.net> Message-ID: <200906041615.16453.yivzhenko@mksat.net> On Thursday 04 June 2009 15:54:24 Rupa Schomaker wrote: > The API mthod of setting the callerid is really for testing purposes. > Since you have no callerid from the FS command line interface I added a > parameter you can pass to specify the CID. > > When in the dialplan you actually have the CID, so none is used. > > mod_lcr is just grabbing the callerid on the channel. I assme this is not > the base callerid you want passed to the sip provider? > Yes, there is internal callerid on the channel. I have external (PSTN) number, associated with this internal number. And i need to pass them to provider. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/54289581/attachment.html From durk.debeer at isp.solcon.nl Thu Jun 4 02:53:05 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Thu, 04 Jun 2009 11:53:05 +0200 Subject: [Freeswitch-users] Attendant transfer problem with a Cisco phone and Freeswitch. Message-ID: I've come across a problem when using Cisco phones as sip-clients on a Freeswitch system. The problem that arises has to do whit which RFC the Cisco phones are following. Best thing to do at this moment is to point to the Tech-invite website http://www.tech-invite.com/. The examples on that site are more explanatory to the problem than I am ever are able to provide. Ok if you take a look at the link SIP Service Examples there are 19 examples of how RFC 5359 describes how call signalling should flow. If you take a careful look at example 5 (attendant transfer) you will discover that before there is the transfer the station the transfer will go to is put on hold. Bob is transfering to Carol so she is being put on hold by Bob, signals 12 to 14. Now if you take a look at the RFC involved in transfering calls, to be found on the main site link SIP-Topics (to the left) and then following the link Call Transfer (middle window second line in yellow section) you'll find three RFC regarding to transfering calls. In none of these am I able to find this putting on hold as is scribed in RFC 5359 so it seems reasonably to assume that this 'putting on hold' in not mandatory. Here now arises the source of my problem, the Cisco phones are using RFC 5359 when attempting a attendant transfer. When now this signalling is to flow through Freeswitch it puts the station where the transfer is going to on hold as in the prior example is happening to Carol. Freeswitch the connects the MOH stream to this station. Seems a logical thing to do if your assume the putting on hold is not mandatory for the transfer. When now the original station (prior example Bob) is sending the refer thins go bad. Freeswitch is not sending new invites or other signaling to the stations. It is only processing the byes from the station that preforms the transfer (prior example Bob). If I now break the MOH stream on the freeswitch cli all goes well meaning that all invites and other signalling is flowing correctly through Freeswitch. Did anyone out there have the same problem or better yet have a fix for it?. ? Kind regards, Durk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/af8949b7/attachment.html From rupa at rupa.com Thu Jun 4 06:57:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 4 Jun 2009 08:57:52 -0500 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: <200906041615.16453.yivzhenko@mksat.net> References: <200906041402.43704.yivzhenko@mksat.net> <200906041615.16453.yivzhenko@mksat.net> Message-ID: Update to at least rev 13611 and mod_lcr will check "effective_caller_id_number" prior to the real callerid on the channel. This works in the same way that bridge does so no surprises. On Thu, Jun 4, 2009 at 8:15 AM, Yuriy Ivzhenko wrote: > On Thursday 04 June 2009 15:54:24 Rupa Schomaker wrote: > > > The API mthod of setting the callerid is really for testing purposes. > > > Since you have no callerid from the FS command line interface I added a > > > parameter you can pass to specify the CID. > > > > > > When in the dialplan you actually have the CID, so none is used. > > > > > > mod_lcr is just grabbing the callerid on the channel. I assme this is not > > > the base callerid you want passed to the sip provider? > > > > > Yes, there is internal callerid on the channel. I have external (PSTN) > number, associated with this internal number. And i need to pass them to > provider. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/6e38ee9a/attachment.html From brian at freeswitch.org Thu Jun 4 07:07:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:07:26 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: <079FC3ED-1F71-4257-A651-FCA66B7C8EB4@freeswitch.org> Yah make sure your ext-rtp-ip is set or hop on IRC and let me help you. /b On Jun 4, 2009, at 4:41 AM, Jason White wrote: > If I apply this to the external profile (using it on the internal > profile has > no effect), the Via headers now receive the external IP address, > i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in > it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if > necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 > 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/64944d25/attachment-0001.html From bruce.mcalister at blueface.ie Thu Jun 4 07:10:11 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Thu, 04 Jun 2009 15:10:11 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <6297FD9D-142A-4E58-B4AC-6294652EAAF6@jerris.com> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> <20090603101746.GA3854@jdc.jasonjgw.net> <4A266025.1070505@blueface.ie> <4A26D26C.8080107@blueface.ie> <6297FD9D-142A-4E58-B4AC-6294652EAAF6@jerris.com> Message-ID: <4A27D5C3.4060209@blueface.ie> Hi Mike, As per request, here is the build status of 1.0.4preX for me: 1.0.4pre2 compile successful 1.0.4pre3 compile successful 1.0.4pre4 compile successful 1.0.4pre5 compile successful 1.0.4pre6 compile successful 1.0.4pre7 compile successful 1.0.4pre8 compile fails I have used the same spec file for each build so the build environment is identical for each. I have filed a jira at: http://jira.freeswitch.org/browse/FSBUILD-169 Thanks Bruce Michael Jerris wrote: > if you could nail down a specific svn revision that causes this issue > and file a jira at http://jira.freeswitch.org that would be a big help > in resolving this issue. > > Mike > > On Jun 3, 2009, at 3:43 PM, Bruce McAlister wrote: > >> Hi All, >> >> Any pointers or suggestions on this issue would be greatly >> appreciated. >> >> PS: I tried compiling several version of FreeSWITCH to see if I >> encounter the same issue, I have varying results. Version 1.0 compiles >> fine, version 1.0.2 fails and 1.0.3RC1/1.0.3 builds fine. But this >> error >> message is new in version 1.0.4preX. I've not tried any older >> pre-releases of 1.0.4 though. >> >> Thanks >> Bruce >> >> Thanks >> Bruce >>> /bin/bash >>> /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> libtool >>> --silent --mode=compile /usr/sfw/bin/gcc -m32 -DHAVE_CONFIG_H >>> -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT >>> -D_LARGEFILE64_SOURCE -I./include >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> include/arch/unix >>> -I./include/arch/unix >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> include >>> -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch >>> locks/unix/thread_mutex.lo >>> In file included from >>> /usr/sfw/lib/gcc/i386-pc-solaris2.10/3.4.3/include/sys/types.h:27, >>> from ./include/apr.h:113, >>> from >>> /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> include/arch/unix/apr_arch_thread_mutex.h:24, >>> from locks/unix/thread_mutex.c:17: >>> /usr/include/sys/feature_tests.h:336:2: #error "Compiler or options >>> invalid; UNIX 03 and POSIX.1-2001 applications require the use >>> of c99" >>> make[2]: *** [locks/unix/thread_mutex.lo] Error 1 >>> make[2]: Leaving directory >>> `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' >>> make[1]: *** [all-recursive] Error 1 >>> make[1]: Leaving directory >>> `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' >>> make: *** [libs/apr/libapr-1.la] Error 2 >>> >>> In all cases I have started the build from the beginning, whereby I >>> remove and re-extract the 1.0.4pre8 tarball. I've tried with just a >>> configure and also a bootstrap/configure, but I end up with the same >>> error (except when I change the compiler to Sun Studio 12's c99). >>> >>> Is GCC 3.4.3 too old to use to build this version of freeswitch? >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Jun 4 07:13:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:13:29 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <8548096.287831244101967346.JavaMail.root@zimbra.tng.de> References: <8548096.287831244101967346.JavaMail.root@zimbra.tng.de> Message-ID: Sorry but this type of trace is impossible to read. I want raw pcap if possible. /b On Jun 4, 2009, at 2:52 AM, Rudolf Denert wrote: > Ok, > > here is the SIP trace. If you need more, just tell me and I will > send them. The RTP trace you already have, haven't you? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3621de5a/attachment.html From brian at freeswitch.org Thu Jun 4 07:16:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:16:20 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: <6DE16413-98FF-4BC3-8067-A7C27081D989@freeswitch.org> The private address in the SDP's o= line is fine. If the far end is rejecting that then something is broken on their side. The c= line is all that matters in this case... For the record we do provide the complete correct SDP to sofia but the lib overrides the o= line and replaces it. I have narrowed down the exact lines of code that causes this in soa.c in sofia... I have emailed the author of the lib to ask why and how to keep sofia from messing with that o= line. /b On Jun 4, 2009, at 4:41 AM, Jason White wrote: > If I apply this to the external profile (using it on the internal > profile has > no effect), the Via headers now receive the external IP address, > i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in > it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if > necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 > 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/03c342ba/attachment.html From yivzhenko at mksat.net Thu Jun 4 07:21:13 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 4 Jun 2009 17:21:13 +0300 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: References: <200906041402.43704.yivzhenko@mksat.net> <200906041615.16453.yivzhenko@mksat.net> Message-ID: <200906041721.13791.yivzhenko@mksat.net> Impressive!!! :) On Thursday 04 June 2009 16:57:52 Rupa Schomaker wrote: > Update to at least rev 13611 and mod_lcr will check > "effective_caller_id_number" prior to the real callerid on the channel. > This works in the same way that bridge does so no surprises. > > On Thu, Jun 4, 2009 at 8:15 AM, Yuriy Ivzhenko wrote: > > On Thursday 04 June 2009 15:54:24 Rupa Schomaker wrote: > > > The API mthod of setting the callerid is really for testing purposes. > > > > > > Since you have no callerid from the FS command line interface I added a > > > > > > parameter you can pass to specify the CID. > > > > > > > > > > > > When in the dialplan you actually have the CID, so none is used. > > > > > > > > > > > > mod_lcr is just grabbing the callerid on the channel. I assme this is > > > not > > > > > > the base callerid you want passed to the sip provider? > > > > Yes, there is internal callerid on the channel. I have external (PSTN) > > number, associated with this internal number. And i need to pass them to > > provider. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/f6b62921/attachment-0001.html From god.nirvana at gmail.com Thu Jun 4 07:19:54 2009 From: god.nirvana at gmail.com (god.nirvana) Date: Thu, 4 Jun 2009 22:19:54 +0800 Subject: [Freeswitch-users] busy tone detect issue Message-ID: <200906042219500623349@gmail.com> hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 openzap.conf: [span zt FXO1] name => OpenZAP-FXO1 number => 1 fxo-channel => 1 [span zt FXO2] name => OpenZAP-FXO2 number => 2 fxo-channel => 2 [span zt FXO3] name => OpenZAP-FXO3 number => 3 fxo-channel => 3 [span zt FXO4] name => OpenZAP-FXO4 number => 4 fxo-channel => 4 openzap.conf.xml : when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/47dd5d21/attachment.html From regs at kinetix.gr Thu Jun 4 07:32:22 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 04 Jun 2009 17:32:22 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior Message-ID: <4A27DAF6.30005@kinetix.gr> In the process of trying to use Freeswitch in a production environment I conducted a number of performance tests using various servers. It was then that I noticed some strange behavior from FS. When I stripped down the scenario I was using to a simple bridge scenario, I stumbled upon a strange behavior. The scenario as I stated is quite simple. |---------| |---------| | |------- Call from sipp------> | | | sipp | | FS | | | <------ Call back to sipp----| | |---------| |---------| I did not use an RTP stream for my calls just to test the signaling alone. The sipp scenario is the standard uac.xml scenario that can be found integrated to sipp with the following options : Test FS 1: sipp :5060 -s 55555555 -i -mi -ci -r 10 -d 5000 -l 100 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~(35-60) % (35 during the generation of new calls, 60 during the -l limit imposed by the test) Note : 985 of them had a duration (billsec) of 10 and 15 of them had a duration of 11. I tried raising the call rate and limit... Test FS 1: sipp :5060 -s 55555555 -i -mi -ci -r 20 -d 5000 -l 200 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~0-30 % (0 during the generation of new calls, 30 during the -l limit imposed by the test) THIS IS WHAT MAKES ME WONDER : The distribution of the durations (billsec - not complete durations) : 183 calls with 10 secs billed duration 110 calls with 11 secs billed duration 238 calls with 12 secs billed duration 447 calls with 13 secs billed duration 22 calls with 14 secs billed duration The sipp scenario is simple "hangup the phone after 10 secs". So, why am I seeing these? Of course that has something to do with the stress the machine has been put through during the second test. But I can see it happening to less stressful conditions (i.e. 15 calls per second) to a smaller extend. I captured one of these calls and verified that when the sipp client hangs up exactly 10 secs after the call start, FS receives the BYE and replies with 200 OK. BUT it does not hang the second leg in a timely manner i.e. it sends a BYE to the sipp server side 1-4 seconds AFTER that. That explains the 11, 12, 13, 14 secs durations seen on the second test. What is more interesting is that I would expect to see in the CDRs the first and second leg to have different durations (since the a leg BYE was received and aknowledged by FS in the correct time) i.e. 10 and 14 secs accordingly. But what I get is the same duration for both legs (14 secs). This in my opinion is very dangerous on production environments (you get charged by your provider more seconds that you charge your clients - or - you falsely charge your clients with bigger durations although they hunged up corectly (and you acknowledged it)). NOTE No 1 : All the performance recommendations found in the wiki has been applied. In fact only the essential modules that could make this scenario work were loaded. NOTE No 2 : I tried using asterisk (as a point of reference - don't get me wrong - I am not trying to start a flame war here). And it succeeded doing on the same hardware 60 calls/sec with a channel limit of 400 sim. calls using only 50% of the cpu (maximum). No under any circumstances I have seen the behavior above (this inability to hang call legs in a timely manner). Even when I pushed asterisk to the limits (80 calls per second 600 max call limit) and it started failing on some calls it never failed to hangup the calls for both legs on exactly 10 secs. NOTE No 3 : As you can tell I was using a very small machine for my tests. When I moved the same tests to larger installations (Quad Core Opterons and Xeons) I got proportional results to the above. NOTE No 4 : The tests were performed in a LAN environment and since there was no RTP involved I think there were no bandwidth issues there. NOTE No 5 : The tests were performed using numerous SVN versions (latest : 13610), the stable version and the 1.0.4pre8 version. NOTE No 6 : Using the -hp switch made no noticeable change in behavior. I am not trying to complain for FS's performance (far from it). I am just somewhat disappointed seeing it performing in such a strange manner when under stress. I would prefer a design that drops the calls after a certain threshold than a design that incorrectly handles them all (I am aware of the max sessions per second in switch.conf.cml - but I am starting to see this behavior even with the cpu idling at about 80%). I don't know if anyone else had the same experience when testing Freeswitch. I can happily supply with all the test details (config files, captures etc) to all interested parties. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From brian at freeswitch.org Thu Jun 4 07:40:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:40:59 -0500 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A27DAF6.30005@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> Message-ID: <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> On Jun 4, 2009, at 9:32 AM, Apostolos Pantsiopoulos wrote: > NOTE No 1 : All the performance recommendations found in the wiki has > been applied. In fact only the essential modules that could make this > scenario work > were loaded. What are you testing against? What OS, Hardware, Distro and such? > NOTE No 2 : I tried using asterisk (as a point of reference - don't > get > me wrong - I am not trying to start a flame war here). And it > succeeded > doing on the same hardware 60 calls/sec with a channel limit of 400 > sim. calls using only 50% of the cpu (maximum). No under any > circumstances I have seen the behavior above (this inability to hang > call legs in a timely manner). Even when I pushed asterisk to the > limits > (80 calls per second 600 max call limit) and it started failing on > some > calls it never failed to hangup the calls for both legs on exactly > 10 secs. Load testing is a science and you can do it wrong most of the time unless you know exactly what you're doing. If you're going against the default dialplan its heavy and not something I would load test against. > NOTE No 3 : As you can tell I was using a very small machine for my > tests. When I moved the same tests to larger installations (Quad Core > Opterons and Xeons) I got proportional results to the above. What are you testing on now? Hope its 64bit. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3389eb15/attachment-0001.html From regs at kinetix.gr Thu Jun 4 08:01:54 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 04 Jun 2009 18:01:54 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> Message-ID: <4A27E1E2.6070905@kinetix.gr> Brian West wrote: > > On Jun 4, 2009, at 9:32 AM, Apostolos Pantsiopoulos wrote: > >> NOTE No 1 : All the performance recommendations found in the wiki has >> been applied. In fact only the essential modules that could make this >> scenario work >> were loaded. > > What are you testing against? What OS, Hardware, Distro and such? The small server tests were made on a 5-year old PC (32 bit, 3 Ghz P4, Cetnos 5.3). The large server 1 : Quad-Core AMD Opteron(tm) Processor 2350 HE (64 bit, Centos 5.3) The large server 2 : Dual-Core AMD Opteron(tm) Processor 2214 HE (64 bit, Centos 5.3) The large server 3 : Intel(R) Xeon(R) CPU E5345 @ 2.33GHz (32 bit, Centos 5.3) > >> NOTE No 2 : I tried using asterisk (as a point of reference - don't get >> me wrong - I am not trying to start a flame war here). And it succeeded >> doing on the same hardware 60 calls/sec with a channel limit of 400 >> sim. calls using only 50% of the cpu (maximum). No under any >> circumstances I have seen the behavior above (this inability to hang >> call legs in a timely manner). Even when I pushed asterisk to the limits >> (80 calls per second 600 max call limit) and it started failing on some >> calls it never failed to hangup the calls for both legs on exactly 10 >> secs. > > Load testing is a science and you can do it wrong most of the time > unless you know exactly what you're doing. If you're going against the > default dialplan its heavy and not something I would load test against. The dialplan : I think it is the simplest that can be used in this scenario. > > >> NOTE No 3 : As you can tell I was using a very small machine for my >> tests. When I moved the same tests to larger installations (Quad Core >> Opterons and Xeons) I got proportional results to the above. > > What are you testing on now? Hope its 64bit. Most of the platforms were 64 bit (although the results that I posted were from the small 32-bit server, the results from the 64-bit servers were proportional to those). In other words we needed a large call/sec rate for the high end servers but in any case the same phenomenon occured at around 60% idle cpu. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From msc at freeswitch.org Thu Jun 4 09:38:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 09:38:01 -0700 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A27E1E2.6070905@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> Message-ID: <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> > > > The dialplan : > > > > > > > > You forgot the parens around .* It should be expression="^(.*)$" if you plan to use $1 later in the extension... > > > data="absolute_codec_string=PCMA"/> > data="sofia/gateway/sipp01/$1"/> ... like here ^^^^^^^ :) -MC > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/a30ef693/attachment.html From evilla at chipoly.com Thu Jun 4 09:45:32 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Thu, 4 Jun 2009 10:45:32 -0600 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup Message-ID: <01ac01c9e533$e0b79430$a226bc90$@com> Hello guys, I have the A400 card and I'm using: wanpipe-3.5.2 driver zaptel-1.4.11 oslec-0.2 freeswitch 1.0.3 CentOS 5.2 x86_64 kernel 2.6.18-12.128.1.10el5 I had my system working with no problems. After I did shutdown/restart, there was a problem loading wanrouter. Please look at this error log: http://pastebin.freeswitch.org/9246 Then I did reinstal zaptel and oslec again, and then wanrouter got to start with no problem. The only issue was that freeswitch got core dumps when starting, so could not get working again the system. For testing purposes I shutdown / restart the box again and got the same error... Any ideas where to look at? ChiPoLy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/7037385c/attachment.html From brian at freeswitch.org Thu Jun 4 09:48:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 11:48:37 -0500 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup In-Reply-To: <01ac01c9e533$e0b79430$a226bc90$@com> References: <01ac01c9e533$e0b79430$a226bc90$@com> Message-ID: Please report your bugs to jira.freeswitch.org... if you have a segfault then thats where it belongs. Thanks, Brian On Jun 4, 2009, at 11:45 AM, Ing. Edwin Villarreal wrote: > Then I did reinstal zaptel and oslec again, and then wanrouter got > to start with no problem. The only issue was that freeswitch got > core dumps when starting, so could not get working again the system. > > For testing purposes I shutdown / restart the box again and got the > same > error... > > Any ideas where to look at? > > ChiPoLy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/1017b04f/attachment.html From larclap at yahoo.com Thu Jun 4 09:59:00 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 4 Jun 2009 09:59:00 -0700 Subject: [Freeswitch-users] Missing lines copying data from console to vi Message-ID: <079f01c9e535$c1fed2e0$45fc78a0$@com> I want to copy the results of a siptrace captured on the fs console to a file. The console is running on a Gnome terminal. I highlight the text I want to copy in the fs console, open a vi session in insert mode, and paste the text. However, the text is not pasted as I copied it - it is missing characters/lines. I know I am doing something wrong. Is there another way to save siptraces to a file? Redirection doesn't work. sofia profile internal siptrace on is the command I use. Thanks Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/9addf86b/attachment.html From evilla at chipoly.com Thu Jun 4 10:03:47 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Thu, 4 Jun 2009 11:03:47 -0600 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup In-Reply-To: References: <01ac01c9e533$e0b79430$a226bc90$@com> Message-ID: <01e901c9e536$6d8aafd0$48a00f70$@com> Thanks Brian. After commenting mod_openzap in modules.conf.xml I get to run FS with no problem. Problem is with Zaptel / Wanpipe interface. L Ing. Edwin Villarreal World Net Commerce SA CV De: Brian West [mailto:brian at freeswitch.org] Enviado el: jueves, 04 de junio de 2009 10:49 a.m. Para: freeswitch-users at lists.freeswitch.org Asunto: Re: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup Please report your bugs to jira.freeswitch.org... if you have a segfault then thats where it belongs. Thanks, Brian On Jun 4, 2009, at 11:45 AM, Ing. Edwin Villarreal wrote: Then I did reinstal zaptel and oslec again, and then wanrouter got to start with no problem. The only issue was that freeswitch got core dumps when starting, so could not get working again the system. For testing purposes I shutdown / restart the box again and got the same error... Any ideas where to look at? ChiPoLy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/f6fb9b3e/attachment-0001.html From brian at freeswitch.org Thu Jun 4 10:06:07 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 12:06:07 -0500 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup In-Reply-To: <01e901c9e536$6d8aafd0$48a00f70$@com> References: <01ac01c9e533$e0b79430$a226bc90$@com> <01e901c9e536$6d8aafd0$48a00f70$@com> Message-ID: If FreeSWITCH segfaulted its a bug we should fix it... we have no carpet so its impossible to sweet things under the rug.. please report the crash to jira. /b On Jun 4, 2009, at 12:03 PM, Ing. Edwin Villarreal wrote: > Thanks Brian. > > After commenting mod_openzap in modules.conf.xml I get to run FS > with no problem. > > Problem is with Zaptel / Wanpipe interface. > > L > > Ing. Edwin Villarreal > World Net Commerce SA CV > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3dbe7fa1/attachment.html From regs at kinetix.gr Thu Jun 4 10:47:21 2009 From: regs at kinetix.gr (regs at kinetix.gr) Date: Thu, 04 Jun 2009 20:47:21 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> Message-ID: <4A2808A9.8070409@kinetix.gr> Michael Collins wrote: > > > The dialplan : > > > > > > > > expression="^.*$"> > > > You forgot the parens around .* > It should be expression="^(.*)$" if you plan to use $1 later in the > extension... > > > > > data="absolute_codec_string=PCMA"/> > data="sofia/gateway/sipp01/$1"/> > > ... like here ^^^^^^^ > :) > -MC You are right! Although, I don't think that would change the outcome of my test :) > > > > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Thu Jun 4 14:12:47 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 16:12:47 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: While the o= line shouldn't matter we have fixed sofia to honor the info we give it and correctly give the right ip in that line also. So please update to 13621 and see if that corrects your issue. Thanks, Brian On Jun 4, 2009, at 4:41 AM, Jason White wrote: > Michael Jerris wrote: >> Can you please re-test with current svn trunk. we added some new nat >> busting code yesterday that may assist with this. You will need to >> specify the new > value="localnet.auto"/ >>> param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 >> ) > > If I apply this to the external profile (using it on the internal > profile has > no effect), the Via headers now receive the external IP address, > i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in > it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if > necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 > 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b9541d22/attachment.html From tayeb.meftah at gmail.com Thu Jun 4 14:34:04 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 21:34:04 +0000 Subject: [Freeswitch-users] WikiPBX Installation Message-ID: <4A283DCC.5040701@gmail.com> hello my friends please i need any Web FrontEnd for FS bicose i'm blind i'm using a screen reader. this screen reader is for windows only also is don't support a consol interface only graphical interface and i have a CentOs4.7 Box that contin my FS installed but i need to manage it (extentions, gateways and other fiturs please anyone setup WikiPBX for me? any help is welcome thanks my friends From matthew.lockwood at gmail.com Thu Jun 4 14:39:30 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 14:39:30 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A283DCC.5040701@gmail.com> References: <4A283DCC.5040701@gmail.com> Message-ID: <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> I'm about to start creating one. I think FS needs a UI comparable to FreePBX or similar. It's the next project on my list. It would certainly be a boost to the FS project and user acceptance. On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: > hello my friends > > please i need any Web FrontEnd for FS > bicose i'm blind i'm using a screen reader. > this screen reader is for windows only > also is don't support a consol interface only graphical interface > and i have a CentOs4.7 Box that contin my FS installed > but i need to manage it (extentions, gateways and other fiturs > please anyone setup WikiPBX for me? > any help is welcome > thanks my friends > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ba232a78/attachment.html From brian at freeswitch.org Thu Jun 4 14:45:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 16:45:08 -0500 Subject: [Freeswitch-users] IPv6 registration fails under rev. 13606 In-Reply-To: <20090604062323.GA19740@jdc.jasonjgw.net> References: <20090604053416.GA9751@jdc.jasonjgw.net> <971C091E-6FEC-471A-855B-66FC976D4E2C@freeswitch.org> <20090604062323.GA19740@jdc.jasonjgw.net> Message-ID: <66DC522C-1874-44C1-9D76-8AF8536CC42A@freeswitch.org> Jason, Can I appoint you the official ipv6 tester? :) Anyway update to trunk and see if everything is ok still. Thanks, Brian On Jun 4, 2009, at 1:23 AM, Jason White wrote: > Brian West wrote: >> A few more checks went in try 13610... See I knew I would introduce >> some >> regressions. :( > > Works for me, thanks! > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/02213d61/attachment-0001.html From tayeb.meftah at gmail.com Thu Jun 4 14:56:13 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 21:56:13 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> Message-ID: <4A2842FD.304@gmail.com> hello, please: * in each programing language you want to start this project? * how i can contribute to it? thanks Matthew Lockwood wrote: > I'm about to start creating one. I think FS needs a UI comparable to > FreePBX or similar. It's the next project on my list. It would > certainly be a boost to the FS project and user acceptance. > > On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb > wrote: > > hello my friends > > please i need any Web FrontEnd for FS > bicose i'm blind i'm using a screen reader. > this screen reader is for windows only > also is don't support a consol interface only graphical interface > and i have a CentOs4.7 Box that contin my FS installed > but i need to manage it (extentions, gateways and other fiturs > please anyone setup WikiPBX for me? > any help is welcome > thanks my friends > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/cc214931/attachment.html From matthew.lockwood at gmail.com Thu Jun 4 15:02:10 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:02:10 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A2842FD.304@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> Message-ID: <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> This may raise eyebrows, but I was thinking about using the dotnet/mono framework. I'm coming from 10+ years in Windows, and have blindly followed MS technologies through my career, so I'm very comfortable with C#. There are inherent issues with this approach though. The other alternative is PHP, although it's not my first choice and I'm nowhere near as comfortable with it. What're your thoughts? On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: > hello, > please: > > - in each programing language you want to start this project? > - how i can contribute to it? > > thanks > Matthew Lockwood wrote: > > I'm about to start creating one. I think FS needs a UI comparable to > FreePBX or similar. It's the next project on my list. It would certainly be > a boost to the FS project and user acceptance. > > On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: > >> hello my friends >> >> please i need any Web FrontEnd for FS >> bicose i'm blind i'm using a screen reader. >> this screen reader is for windows only >> also is don't support a consol interface only graphical interface >> and i have a CentOs4.7 Box that contin my FS installed >> but i need to manage it (extentions, gateways and other fiturs >> please anyone setup WikiPBX for me? >> any help is welcome >> thanks my friends >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/fb539e22/attachment.html From tayeb.meftah at gmail.com Thu Jun 4 15:05:46 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:05:46 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> Message-ID: <4A28453A.6040205@gmail.com> hello i'm also a windows Developer / microsoft technologie user i use.Net/Mono/... #Develope, SQLite and ... are you in irc now? my nick: DelphiWorld please message me. thanks Matthew Lockwood wrote: > This may raise eyebrows, but I was thinking about using the > dotnet/mono framework. I'm coming from 10+ years in Windows, and have > blindly followed MS technologies through my career, so I'm very > comfortable with C#. There are inherent issues with this approach > though. The other alternative is PHP, although it's not my first > choice and I'm nowhere near as comfortable with it. > > What're your thoughts? > > On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb > wrote: > > hello, > please: > > * in each programing language you want to start this project? > * how i can contribute to it? > > thanks > Matthew Lockwood wrote: >> I'm about to start creating one. I think FS needs a UI comparable >> to FreePBX or similar. It's the next project on my list. It would >> certainly be a boost to the FS project and user acceptance. >> >> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >> > wrote: >> >> hello my friends >> >> please i need any Web FrontEnd for FS >> bicose i'm blind i'm using a screen reader. >> this screen reader is for windows only >> also is don't support a consol interface only graphical interface >> and i have a CentOs4.7 Box that contin my FS installed >> but i need to manage it (extentions, gateways and other fiturs >> please anyone setup WikiPBX for me? >> any help is welcome >> thanks my friends >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/86c240bf/attachment.html From anthony.minessale at gmail.com Thu Jun 4 15:08:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 17:08:35 -0500 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A2808A9.8070409@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> <4A2808A9.8070409@kinetix.gr> Message-ID: <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> FS uses async rtp timers so you may want to set rtp-timer-name=none in the profile param to simulate asterisk conditions. Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit single cpu box because that was what was popular when it was designed and the chance for race conditions is minimal because there is only 1 cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic difference. I will be happy to investigate this issue a bit if you'd like but i do not have any box like you describe so if I can't find anything you may have to lend us your lab. On Thu, Jun 4, 2009 at 12:47 PM, regs at kinetix.gr wrote: > Michael Collins wrote: > > > > > > The dialplan : > > > > > > > > > > > > > > > > > expression="^.*$"> > > > > > > You forgot the parens around .* > > It should be expression="^(.*)$" if you plan to use $1 later in the > > extension... > > > > > > > > > > > data="absolute_codec_string=PCMA"/> > > > data="sofia/gateway/sipp01/$1"/> > > > > ... like here ^^^^^^^ > > :) > > -MC > > You are right! Although, I don't think that would change the outcome of > my test :) > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/459b5035/attachment-0001.html From matthew.lockwood at gmail.com Thu Jun 4 15:09:16 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:09:16 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A28453A.6040205@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> Message-ID: <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> KeeblerElfMatt ... I'll be there in a minute On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: > hello > i'm also a windows Developer / microsoft technologie user > i use.Net/Mono/... #Develope, SQLite and ... > are you in irc now? > my nick: DelphiWorld > please message me. > thanks > Matthew Lockwood wrote: > > This may raise eyebrows, but I was thinking about using the dotnet/mono > framework. I'm coming from 10+ years in Windows, and have blindly followed > MS technologies through my career, so I'm very comfortable with C#. There > are inherent issues with this approach though. The other alternative is PHP, > although it's not my first choice and I'm nowhere near as comfortable with > it. > > What're your thoughts? > > On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: > >> hello, >> please: >> >> - in each programing language you want to start this project? >> - how i can contribute to it? >> >> thanks >> Matthew Lockwood wrote: >> >> I'm about to start creating one. I think FS needs a UI comparable to >> FreePBX or similar. It's the next project on my list. It would certainly be >> a boost to the FS project and user acceptance. >> >> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >> >>> hello my friends >>> >>> please i need any Web FrontEnd for FS >>> bicose i'm blind i'm using a screen reader. >>> this screen reader is for windows only >>> also is don't support a consol interface only graphical interface >>> and i have a CentOs4.7 Box that contin my FS installed >>> but i need to manage it (extentions, gateways and other fiturs >>> please anyone setup WikiPBX for me? >>> any help is welcome >>> thanks my friends >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/12fb4c92/attachment.html From intralanman at freeswitch.org Thu Jun 4 15:16:27 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 04 Jun 2009 18:16:27 -0400 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> Message-ID: <4A2847BB.8020306@freeswitch.org> I think you're more likely to get more help if you write it in PHP or ruby. PHP seems to be the open-source web language of choice by most, but ruby/rails has been gaining a lot of ground, so either would probably be a safe bet. -Ray Matthew Lockwood wrote: > KeeblerElfMatt ... I'll be there in a minute > > On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb > wrote: > > hello > i'm also a windows Developer / microsoft technologie user > i use.Net/Mono/... #Develope, SQLite and ... > are you in irc now? > my nick: DelphiWorld > please message me. > thanks > Matthew Lockwood wrote: >> This may raise eyebrows, but I was thinking about using the >> dotnet/mono framework. I'm coming from 10+ years in Windows, and >> have blindly followed MS technologies through my career, so I'm >> very comfortable with C#. There are inherent issues with this >> approach though. The other alternative is PHP, although it's not >> my first choice and I'm nowhere near as comfortable with it. >> >> What're your thoughts? >> >> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >> > wrote: >> >> hello, >> please: >> >> * in each programing language you want to start this project? >> * how i can contribute to it? >> >> thanks >> Matthew Lockwood wrote: >>> I'm about to start creating one. I think FS needs a UI >>> comparable to FreePBX or similar. It's the next project on >>> my list. It would certainly be a boost to the FS project and >>> user acceptance. >>> >>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>> > wrote: >>> >>> hello my friends >>> >>> please i need any Web FrontEnd for FS >>> bicose i'm blind i'm using a screen reader. >>> this screen reader is for windows only >>> also is don't support a consol interface only graphical >>> interface >>> and i have a CentOs4.7 Box that contin my FS installed >>> but i need to manage it (extentions, gateways and other >>> fiturs >>> please anyone setup WikiPBX for me? >>> any help is welcome >>> thanks my friends >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d389a666/attachment.html From tayeb.meftah at gmail.com Thu Jun 4 15:22:20 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:22:20 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A2847BB.8020306@freeswitch.org> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> Message-ID: <4A28491C.7050009@gmail.com> hello yes, PHP/Ruby is the best Web Scripting Languages but each one in the World have there knoledge base limitation thanks Raymond Chandler wrote: > I think you're more likely to get more help if you write it in PHP or > ruby. PHP seems to be the open-source web language of choice by most, > but ruby/rails has been gaining a lot of ground, so either would > probably be a safe bet. > > -Ray > > Matthew Lockwood wrote: >> KeeblerElfMatt ... I'll be there in a minute >> >> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb > > wrote: >> >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >>> This may raise eyebrows, but I was thinking about using the >>> dotnet/mono framework. I'm coming from 10+ years in Windows, and >>> have blindly followed MS technologies through my career, so I'm >>> very comfortable with C#. There are inherent issues with this >>> approach though. The other alternative is PHP, although it's not >>> my first choice and I'm nowhere near as comfortable with it. >>> >>> What're your thoughts? >>> >>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>> > wrote: >>> >>> hello, >>> please: >>> >>> * in each programing language you want to start this >>> project? >>> * how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>>> I'm about to start creating one. I think FS needs a UI >>>> comparable to FreePBX or similar. It's the next project on >>>> my list. It would certainly be a boost to the FS project >>>> and user acceptance. >>>> >>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>> > wrote: >>>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only graphical >>>> interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and other >>>> fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ Freeswitch-users >>> mailing list Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/c771db36/attachment-0001.html From matthew.lockwood at gmail.com Thu Jun 4 15:23:01 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:23:01 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A2847BB.8020306@freeswitch.org> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> Message-ID: <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> There are two areas of help I'll need for sure: 1) Understanding certain nuances of FreeSWITCH and how to make certain things work (there are things I'll need to implement that aren't in my implementation) 2) User testing and lots and lots of feedback. I was planning to initially adopt the FreePBX look. If there's a web designer that wants to put together a new interface, that would most certainly be a welcome addition. On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler wrote: > I think you're more likely to get more help if you write it in PHP or > ruby. PHP seems to be the open-source web language of choice by most, but > ruby/rails has been gaining a lot of ground, so either would probably be a > safe bet. > > -Ray > > Matthew Lockwood wrote: > > KeeblerElfMatt ... I'll be there in a minute > > On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: > >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >> >> This may raise eyebrows, but I was thinking about using the dotnet/mono >> framework. I'm coming from 10+ years in Windows, and have blindly followed >> MS technologies through my career, so I'm very comfortable with C#. There >> are inherent issues with this approach though. The other alternative is PHP, >> although it's not my first choice and I'm nowhere near as comfortable with >> it. >> >> What're your thoughts? >> >> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >> >>> hello, >>> please: >>> >>> - in each programing language you want to start this project? >>> - how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>> >>> I'm about to start creating one. I think FS needs a UI comparable to >>> FreePBX or similar. It's the next project on my list. It would certainly be >>> a boost to the FS project and user acceptance. >>> >>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only graphical interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and other fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/41025f7f/attachment.html From matthew.lockwood at gmail.com Thu Jun 4 15:25:03 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:25:03 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A28491C.7050009@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <4A28491C.7050009@gmail.com> Message-ID: <415541b10906041525l25b6cf3ak56d390e113b0fc9d@mail.gmail.com> Ray - The other issue is my competence in those scripting languages; I'm very competent in C# and the .NET framework. I'll have a substantial amount of downtime while I'm on several long-haul flights in the near future, so I can devote a good amount of time to getting this done. Mefta - I'm in IRC right now. On Thu, Jun 4, 2009 at 3:22 PM, Meftah Tayeb wrote: > hello > yes, PHP/Ruby is the best Web Scripting Languages > but each one in the World have there knoledge base limitation > thanks > Raymond Chandler wrote: > > I think you're more likely to get more help if you write it in PHP or ruby. > PHP seems to be the open-source web language of choice by most, but > ruby/rails has been gaining a lot of ground, so either would probably be a > safe bet. > > -Ray > > Matthew Lockwood wrote: > > KeeblerElfMatt ... I'll be there in a minute > > On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: > >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >> >> This may raise eyebrows, but I was thinking about using the dotnet/mono >> framework. I'm coming from 10+ years in Windows, and have blindly followed >> MS technologies through my career, so I'm very comfortable with C#. There >> are inherent issues with this approach though. The other alternative is PHP, >> although it's not my first choice and I'm nowhere near as comfortable with >> it. >> >> What're your thoughts? >> >> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >> >>> hello, >>> please: >>> >>> - in each programing language you want to start this project? >>> - how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>> >>> I'm about to start creating one. I think FS needs a UI comparable to >>> FreePBX or similar. It's the next project on my list. It would certainly be >>> a boost to the FS project and user acceptance. >>> >>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only graphical interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and other fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/12292170/attachment-0001.html From brian at freeswitch.org Thu Jun 4 15:27:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 17:27:55 -0500 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: ACK!~ :P Don't copy, be original... thats how you win! /b On Jun 4, 2009, at 5:23 PM, Matthew Lockwood wrote: > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ecb91011/attachment.html From tayeb.meftah at gmail.com Thu Jun 4 15:28:32 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:28:32 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: <4A284A90.70602@gmail.com> hi, yes, i'm here to help you developing this Web GUI bicose only WikiPBX is here now and is not easy to install i hop that you start one with me thanks Matthew Lockwood wrote: > There are two areas of help I'll need for sure: > 1) Understanding certain nuances of FreeSWITCH and how to make certain > things work (there are things I'll need to implement that aren't in my > implementation) > 2) User testing and lots and lots of feedback. > > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. > > On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler > > wrote: > > I think you're more likely to get more help if you write it in PHP > or ruby. PHP seems to be the open-source web language of choice by > most, but ruby/rails has been gaining a lot of ground, so either > would probably be a safe bet. > > -Ray > > Matthew Lockwood wrote: >> KeeblerElfMatt ... I'll be there in a minute >> >> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb >> > wrote: >> >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >>> This may raise eyebrows, but I was thinking about using the >>> dotnet/mono framework. I'm coming from 10+ years in Windows, >>> and have blindly followed MS technologies through my career, >>> so I'm very comfortable with C#. There are inherent issues >>> with this approach though. The other alternative is PHP, >>> although it's not my first choice and I'm nowhere near as >>> comfortable with it. >>> >>> What're your thoughts? >>> >>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>> > wrote: >>> >>> hello, >>> please: >>> >>> * in each programing language you want to start this >>> project? >>> * how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>>> I'm about to start creating one. I think FS needs a UI >>>> comparable to FreePBX or similar. It's the next project >>>> on my list. It would certainly be a boost to the FS >>>> project and user acceptance. >>>> >>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>> >>> > wrote: >>>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only >>>> graphical interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and >>>> other fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/de20205a/attachment.html From tayeb.meftah at gmail.com Thu Jun 4 15:30:38 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:30:38 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041525l25b6cf3ak56d390e113b0fc9d@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <4A28491C.7050009@gmail.com> <415541b10906041525l25b6cf3ak56d390e113b0fc9d@mail.gmail.com> Message-ID: <4A284B0E.6000900@gmail.com> hello please send to me a private Message (DelphiWorld) i can't find your Nick name thanks Matthew Lockwood wrote: > Ray - The other issue is my competence in those scripting languages; > I'm very competent in C# and the .NET framework. I'll have a > substantial amount of downtime while I'm on several long-haul flights > in the near future, so I can devote a good amount of time to getting > this done. > > Mefta - I'm in IRC right now. > > On Thu, Jun 4, 2009 at 3:22 PM, Meftah Tayeb > wrote: > > hello > yes, PHP/Ruby is the best Web Scripting Languages > but each one in the World have there knoledge base limitation > thanks > Raymond Chandler wrote: >> I think you're more likely to get more help if you write it in >> PHP or ruby. PHP seems to be the open-source web language of >> choice by most, but ruby/rails has been gaining a lot of ground, >> so either would probably be a safe bet. >> >> -Ray >> >> Matthew Lockwood wrote: >>> KeeblerElfMatt ... I'll be there in a minute >>> >>> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb >>> > wrote: >>> >>> hello >>> i'm also a windows Developer / microsoft technologie user >>> i use.Net/Mono/... #Develope, SQLite and ... >>> are you in irc now? >>> my nick: DelphiWorld >>> please message me. >>> thanks >>> Matthew Lockwood wrote: >>>> This may raise eyebrows, but I was thinking about using the >>>> dotnet/mono framework. I'm coming from 10+ years in >>>> Windows, and have blindly followed MS technologies through >>>> my career, so I'm very comfortable with C#. There are >>>> inherent issues with this approach though. The other >>>> alternative is PHP, although it's not my first choice and >>>> I'm nowhere near as comfortable with it. >>>> >>>> What're your thoughts? >>>> >>>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>>> > wrote: >>>> >>>> hello, >>>> please: >>>> >>>> * in each programing language you want to start >>>> this project? >>>> * how i can contribute to it? >>>> >>>> thanks >>>> Matthew Lockwood wrote: >>>>> I'm about to start creating one. I think FS needs a UI >>>>> comparable to FreePBX or similar. It's the next >>>>> project on my list. It would certainly be a boost to >>>>> the FS project and user acceptance. >>>>> >>>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>>> >>>> > wrote: >>>>> >>>>> hello my friends >>>>> >>>>> please i need any Web FrontEnd for FS >>>>> bicose i'm blind i'm using a screen reader. >>>>> this screen reader is for windows only >>>>> also is don't support a consol interface only >>>>> graphical interface >>>>> and i have a CentOs4.7 Box that contin my FS installed >>>>> but i need to manage it (extentions, gateways and >>>>> other fiturs >>>>> please anyone setup WikiPBX for me? >>>>> any help is welcome >>>>> thanks my friends >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ac11fec7/attachment-0001.html From tayeb.meftah at gmail.com Thu Jun 4 15:31:41 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:31:41 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: <4A284B4D.5000004@gmail.com> hello, .Net is no only for windows but Linux also using MONO and #develope is open source thanks Brian West wrote: > ACK!~ :P Don't copy, be original... thats how you win! > > /b > > On Jun 4, 2009, at 5:23 PM, Matthew Lockwood wrote: > >> I was planning to initially adopt the FreePBX look. If there's a web >> designer that wants to put together a new interface, that would most >> certainly be a welcome addition. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/24dd9ecb/attachment.html From msc at freeswitch.org Thu Jun 4 15:34:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 15:34:32 -0700 Subject: [Freeswitch-users] ATTENTION: Outstanding JIRA Issues Message-ID: <87f2f3b90906041534y316eda97m96c98883a1f1ed7b@mail.gmail.com> To all FreeSWITCHers, If you have any open issues on JIRA please tend to them ASAP. If you have any issues that are not yet reported please report them ASAP. If you have bugs you are not sure about please join IRC and ask for assistance. We are pushing very hard to get as many bugs resolved as possible prior to 1.0.4. Thanks for your assistance with making FreeSWITCH such an awesome project with a great community! -Michael http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/815f955f/attachment.html From matthew.lockwood at gmail.com Thu Jun 4 15:34:43 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:34:43 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A284A90.70602@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> Message-ID: <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> An slightly more graphical install wizard would actually be useful too. This could be extended into something that could install and configure FreeSWITCH and install the GUI all at the same time. Talk about lowering the barrier to entry! Brian - you're completely right. This idea has only been floating around in my head for 48 hours so it's not very mature. As we all know, it's 80% marketing, and 20% product (no smack talk, but think Asterisk!). I'll come up with a strategy. On Thu, Jun 4, 2009 at 3:28 PM, Meftah Tayeb wrote: > hi, > yes, i'm here to help you developing this Web GUI > bicose only WikiPBX is here now and is not easy to install > i hop that you start one with me > thanks > Matthew Lockwood wrote: > > There are two areas of help I'll need for sure: > 1) Understanding certain nuances of FreeSWITCH and how to make certain > things work (there are things I'll need to implement that aren't in my > implementation) > 2) User testing and lots and lots of feedback. > > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. > > On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler < > intralanman at freeswitch.org> wrote: > >> I think you're more likely to get more help if you write it in PHP or >> ruby. PHP seems to be the open-source web language of choice by most, but >> ruby/rails has been gaining a lot of ground, so either would probably be a >> safe bet. >> >> -Ray >> >> Matthew Lockwood wrote: >> >> KeeblerElfMatt ... I'll be there in a minute >> >> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: >> >>> hello >>> i'm also a windows Developer / microsoft technologie user >>> i use.Net/Mono/... #Develope, SQLite and ... >>> are you in irc now? >>> my nick: DelphiWorld >>> please message me. >>> thanks >>> Matthew Lockwood wrote: >>> >>> This may raise eyebrows, but I was thinking about using the dotnet/mono >>> framework. I'm coming from 10+ years in Windows, and have blindly followed >>> MS technologies through my career, so I'm very comfortable with C#. There >>> are inherent issues with this approach though. The other alternative is PHP, >>> although it's not my first choice and I'm nowhere near as comfortable with >>> it. >>> >>> What're your thoughts? >>> >>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >>> >>>> hello, >>>> please: >>>> >>>> - in each programing language you want to start this project? >>>> - how i can contribute to it? >>>> >>>> thanks >>>> Matthew Lockwood wrote: >>>> >>>> I'm about to start creating one. I think FS needs a UI comparable to >>>> FreePBX or similar. It's the next project on my list. It would certainly be >>>> a boost to the FS project and user acceptance. >>>> >>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>>> >>>>> hello my friends >>>>> >>>>> please i need any Web FrontEnd for FS >>>>> bicose i'm blind i'm using a screen reader. >>>>> this screen reader is for windows only >>>>> also is don't support a consol interface only graphical interface >>>>> and i have a CentOs4.7 Box that contin my FS installed >>>>> but i need to manage it (extentions, gateways and other fiturs >>>>> please anyone setup WikiPBX for me? >>>>> any help is welcome >>>>> thanks my friends >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/548a5d37/attachment-0001.html From tayeb.meftah at gmail.com Thu Jun 4 15:38:05 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:38:05 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> Message-ID: <4A284CCD.6030806@gmail.com> hellok, do you need a graphical installer? i'm here to create it for the community! please tel me how i start a new FS subproject thanks Matthew Lockwood wrote: > An slightly more graphical install wizard would actually be useful > too. This could be extended into something that could install and > configure FreeSWITCH and install the GUI all at the same time. Talk > about lowering the barrier to entry! > > Brian - you're completely right. This idea has only been floating > around in my head for 48 hours so it's not very mature. As we all > know, it's 80% marketing, and 20% product (no smack talk, but think > Asterisk!). I'll come up with a strategy. > > On Thu, Jun 4, 2009 at 3:28 PM, Meftah Tayeb > wrote: > > hi, > yes, i'm here to help you developing this Web GUI > bicose only WikiPBX is here now and is not easy to install > i hop that you start one with me > thanks > Matthew Lockwood wrote: >> There are two areas of help I'll need for sure: >> 1) Understanding certain nuances of FreeSWITCH and how to make >> certain things work (there are things I'll need to implement that >> aren't in my implementation) >> 2) User testing and lots and lots of feedback. >> >> I was planning to initially adopt the FreePBX look. If there's a >> web designer that wants to put together a new interface, that >> would most certainly be a welcome addition. >> >> On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler >> > >> wrote: >> >> I think you're more likely to get more help if you write it >> in PHP or ruby. PHP seems to be the open-source web language >> of choice by most, but ruby/rails has been gaining a lot of >> ground, so either would probably be a safe bet. >> >> -Ray >> >> Matthew Lockwood wrote: >>> KeeblerElfMatt ... I'll be there in a minute >>> >>> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb >>> > wrote: >>> >>> hello >>> i'm also a windows Developer / microsoft technologie user >>> i use.Net/Mono/... #Develope, SQLite and ... >>> are you in irc now? >>> my nick: DelphiWorld >>> please message me. >>> thanks >>> Matthew Lockwood wrote: >>>> This may raise eyebrows, but I was thinking about using >>>> the dotnet/mono framework. I'm coming from 10+ years in >>>> Windows, and have blindly followed MS technologies >>>> through my career, so I'm very comfortable with C#. >>>> There are inherent issues with this approach though. >>>> The other alternative is PHP, although it's not my >>>> first choice and I'm nowhere near as comfortable with it. >>>> >>>> What're your thoughts? >>>> >>>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>>> >>> > wrote: >>>> >>>> hello, >>>> please: >>>> >>>> * in each programing language you want to start >>>> this project? >>>> * how i can contribute to it? >>>> >>>> thanks >>>> Matthew Lockwood wrote: >>>>> I'm about to start creating one. I think FS needs >>>>> a UI comparable to FreePBX or similar. It's the >>>>> next project on my list. It would certainly be a >>>>> boost to the FS project and user acceptance. >>>>> >>>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>>> >>>> > wrote: >>>>> >>>>> hello my friends >>>>> >>>>> please i need any Web FrontEnd for FS >>>>> bicose i'm blind i'm using a screen reader. >>>>> this screen reader is for windows only >>>>> also is don't support a consol interface only >>>>> graphical interface >>>>> and i have a CentOs4.7 Box that contin my FS >>>>> installed >>>>> but i need to manage it (extentions, gateways >>>>> and other fiturs >>>>> please anyone setup WikiPBX for me? >>>>> any help is welcome >>>>> thanks my friends >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d65b467d/attachment.html From msc at freeswitch.org Thu Jun 4 15:39:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 15:39:13 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> Message-ID: <87f2f3b90906041539r47bdf2c6m1df4671ad4f67342@mail.gmail.com> On Thu, Jun 4, 2009 at 2:39 PM, Matthew Lockwood wrote: > I'm about to start creating one. I think FS needs a UI comparable to > FreePBX or similar. It's the next project on my list. It would certainly be > a boost to the FS project and user acceptance. > Be sure to check with Bougyman on IRC. He is planning on releasing an open-source, MIT-licensed FS-GUI w/ underlying framework. Last I heard he said week of June 15. Note: he said that it will require Ruby plus Rack and PostgreSQL. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/c11fa02e/attachment.html From msc at freeswitch.org Thu Jun 4 15:40:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 15:40:10 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> Message-ID: <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> On Thu, Jun 4, 2009 at 3:34 PM, Matthew Lockwood wrote: > An slightly more graphical install wizard would actually be useful too. > This could be extended into something that could install and configure > FreeSWITCH and install the GUI all at the same time. Talk about lowering the > barrier to entry! > > Brian - you're completely right. This idea has only been floating around in > my head for 48 hours so it's not very mature. As we all know, it's 80% > marketing, and 20% product (no smack talk, but think Asterisk!). I'll come > up with a strategy. well, the numbers are more like 98.5% marketing in that case. :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b3084ccb/attachment-0001.html From matthew.lockwood at gmail.com Thu Jun 4 15:41:32 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:41:32 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A284CCD.6030806@gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <4A284CCD.6030806@gmail.com> Message-ID: <415541b10906041541s3f01355axef05b978a7c1f5bf@mail.gmail.com> That would be incredible of you. Let's lower the barrier to entry to total noobs! The simpler we can make it the more successful the project will be. On Thu, Jun 4, 2009 at 3:38 PM, Meftah Tayeb wrote: > hellok, > do you need a graphical installer? > i'm here to create it for the community! > please tel me how i start a new FS subproject > thanks > Matthew Lockwood wrote: > > An slightly more graphical install wizard would actually be useful too. > This could be extended into something that could install and configure > FreeSWITCH and install the GUI all at the same time. Talk about lowering the > barrier to entry! > > Brian - you're completely right. This idea has only been floating around in > my head for 48 hours so it's not very mature. As we all know, it's 80% > marketing, and 20% product (no smack talk, but think Asterisk!). I'll come > up with a strategy. > > On Thu, Jun 4, 2009 at 3:28 PM, Meftah Tayeb wrote: > >> hi, >> yes, i'm here to help you developing this Web GUI >> bicose only WikiPBX is here now and is not easy to install >> i hop that you start one with me >> thanks >> Matthew Lockwood wrote: >> >> There are two areas of help I'll need for sure: >> 1) Understanding certain nuances of FreeSWITCH and how to make certain >> things work (there are things I'll need to implement that aren't in my >> implementation) >> 2) User testing and lots and lots of feedback. >> >> I was planning to initially adopt the FreePBX look. If there's a web >> designer that wants to put together a new interface, that would most >> certainly be a welcome addition. >> >> On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler < >> intralanman at freeswitch.org> wrote: >> >>> I think you're more likely to get more help if you write it in PHP or >>> ruby. PHP seems to be the open-source web language of choice by most, but >>> ruby/rails has been gaining a lot of ground, so either would probably be a >>> safe bet. >>> >>> -Ray >>> >>> Matthew Lockwood wrote: >>> >>> KeeblerElfMatt ... I'll be there in a minute >>> >>> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: >>> >>>> hello >>>> i'm also a windows Developer / microsoft technologie user >>>> i use.Net/Mono/... #Develope, SQLite and ... >>>> are you in irc now? >>>> my nick: DelphiWorld >>>> please message me. >>>> thanks >>>> Matthew Lockwood wrote: >>>> >>>> This may raise eyebrows, but I was thinking about using the >>>> dotnet/mono framework. I'm coming from 10+ years in Windows, and have >>>> blindly followed MS technologies through my career, so I'm very comfortable >>>> with C#. There are inherent issues with this approach though. The other >>>> alternative is PHP, although it's not my first choice and I'm nowhere near >>>> as comfortable with it. >>>> >>>> What're your thoughts? >>>> >>>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >>>> >>>>> hello, >>>>> please: >>>>> >>>>> - in each programing language you want to start this project? >>>>> - how i can contribute to it? >>>>> >>>>> thanks >>>>> Matthew Lockwood wrote: >>>>> >>>>> I'm about to start creating one. I think FS needs a UI comparable to >>>>> FreePBX or similar. It's the next project on my list. It would certainly be >>>>> a boost to the FS project and user acceptance. >>>>> >>>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>>>> >>>>>> hello my friends >>>>>> >>>>>> please i need any Web FrontEnd for FS >>>>>> bicose i'm blind i'm using a screen reader. >>>>>> this screen reader is for windows only >>>>>> also is don't support a consol interface only graphical interface >>>>>> and i have a CentOs4.7 Box that contin my FS installed >>>>>> but i need to manage it (extentions, gateways and other fiturs >>>>>> please anyone setup WikiPBX for me? >>>>>> any help is welcome >>>>>> thanks my friends >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/521658d2/attachment.html From brian at freeswitch.org Thu Jun 4 15:41:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 17:41:55 -0500 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> Message-ID: <90DA2AF2-1BDF-4FBE-8732-626B31874B6F@freeswitch.org> Its called marchitecture! /b On Jun 4, 2009, at 5:40 PM, Michael Collins wrote: > well, the numbers are more like 98.5% marketing in that case. :) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From matthew.lockwood at gmail.com Thu Jun 4 15:42:48 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:42:48 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> Message-ID: <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> LOL. I'd rather have a mediocre product and excellent marketing than the other way around; building a better mouse trap and sitting back is totally ineffective. A second GUI wouldn't be a bad thing. On Thu, Jun 4, 2009 at 3:40 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 3:34 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> An slightly more graphical install wizard would actually be useful too. >> This could be extended into something that could install and configure >> FreeSWITCH and install the GUI all at the same time. Talk about lowering the >> barrier to entry! >> >> Brian - you're completely right. This idea has only been floating around >> in my head for 48 hours so it's not very mature. As we all know, it's 80% >> marketing, and 20% product (no smack talk, but think Asterisk!). I'll come >> up with a strategy. > > > well, the numbers are more like 98.5% marketing in that case. :) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ef0e66a1/attachment-0001.html From matthew.lockwood at gmail.com Thu Jun 4 15:44:34 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:44:34 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: <415541b10906041544p349b408dg3073ec5cb771194b@mail.gmail.com> You're very, very right. Okay, so I'm going to need a web designer in the community to work with! Given that this idea has only been floating around in my mind for 48 hours, I've barely thought about it. Although this is a FOSS project, I should apply all the rules of business I've been successfully applying for years. On Thu, Jun 4, 2009 at 3:27 PM, Brian West wrote: > ACK!~ :P Don't copy, be original... thats how you win! > /b > > On Jun 4, 2009, at 5:23 PM, Matthew Lockwood wrote: > > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/5e891335/attachment.html From brian at freeswitch.org Thu Jun 4 15:46:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 17:46:37 -0500 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> Message-ID: <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> I could agree the community needs to come together and JUST DO IT... but I can't count the times I have seen this thread before. /b On Jun 4, 2009, at 5:42 PM, Matthew Lockwood wrote: > LOL. I'd rather have a mediocre product and excellent marketing than > the other way around; building a better mouse trap and sitting back > is totally ineffective. > > A second GUI wouldn't be a bad thing. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0fcb2889/attachment.html From matthew.lockwood at gmail.com Thu Jun 4 15:48:09 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:48:09 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> References: <4A283DCC.5040701@gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> Message-ID: <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> Okay, I'm going to schedule in a start date of the 16th. This will have my company backing it so it will be done! On Thu, Jun 4, 2009 at 3:46 PM, Brian West wrote: > I could agree the community needs to come together and JUST DO IT... but I > can't count the times I have seen this thread before. > /b > > On Jun 4, 2009, at 5:42 PM, Matthew Lockwood wrote: > > LOL. I'd rather have a mediocre product and excellent marketing than the > other way around; building a better mouse trap and sitting back is totally > ineffective. > > A second GUI wouldn't be a bad thing. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3a7d7d36/attachment.html From matthew.lockwood at gmail.com Thu Jun 4 16:13:53 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 16:13:53 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> Message-ID: <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> Okay, leave this with me - I'll bring this together and do what's required. I'll see if I can squeeze it in ahead of schedule, but don't count on it. Also expect pleas for help at some stage, and I'll need a UI developer to create an awesome interface. You'll all be hearing from me shortly! M On Thu, Jun 4, 2009 at 3:48 PM, Matthew Lockwood wrote: > Okay, I'm going to schedule in a start date of the 16th. This will have my > company backing it so it will be done! > > On Thu, Jun 4, 2009 at 3:46 PM, Brian West wrote: > >> I could agree the community needs to come together and JUST DO IT... but I >> can't count the times I have seen this thread before. >> /b >> >> On Jun 4, 2009, at 5:42 PM, Matthew Lockwood wrote: >> >> LOL. I'd rather have a mediocre product and excellent marketing than the >> other way around; building a better mouse trap and sitting back is totally >> ineffective. >> >> A second GUI wouldn't be a bad thing. >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b6cbc6a2/attachment.html From jim at evolutiontel.net Thu Jun 4 16:15:08 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 09:15:08 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> Message-ID: Hi Anthony, Traces as requested. Let me know if you want a jira opened or any further data. Regards, Jim On Thu, Jun 4, 2009 at 10:59 PM, Anthony Minessale wrote: > you should have also turned in the sip trace > sofia profile internal siptrace on > > > On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke wrote: >> >> Hi All, >> >> Looking for some debugging tips and comments on what might be causing >> the media port in the 200OK ( Answer message) to be set to 0 by >> freeswitch. ?Essentially it looks like data might be getting trampled >> somehow. >> >> Portion of 200OK going into Freeswitch >> m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 >> NSE/8000..a= >> ?fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >> 0-15..a=ptime:20..a=sendrecv.. >> >> Portion of 200OK coming out of Freeswitch >> m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 >> NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event >> ?/8000..a=fmtp:101 0-15.. >> >> Note the media port has been set to 0 and the rtpmap for G729 is also >> not correct. ?On receipt of this bad 200Ok the originator sends a BYE. >> >> We are using FS as a B2BUA with bypass_media set to true. ?Thus IMHO >> Freeswitch should not be touching the SDP portion of the message and >> just passing it through. >> >> This can reproduce this at will, so I can collect whatever data is >> nessicary. ?I have added the sofia debug from the console. >> >> Thanks, >> -- >> Jim Burke >> Director Evolutiontel. >> http://www.evolutiontel.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net -------------- next part -------------- login as: root root at 202.xxx.xx.xx's password: Last login: Thu Jun 4 21:56:36 2009 from 60-241-91-137.static.tpgi.com.au [root at sip01 ~]# cd /usr/local/freeswitch/bin [root at sip01 bin]# ./fs_cli _____ ____ ____ _ ___ | ___/ ___| / ___| | |_ _| | |_ \___ \ | | | | | | | _| ___) | | |___| |___ | | |_| |____/ \____|_____|___| ***************************************************** * Anthony Minessale II, Ken Rice, Michael Jerris * * FreeSWITCH (http://www.freeswitch.org) * * Brought to you by ClueCon http://www.cluecon.com/ * ***************************************************** Type /help to see a list of commands +OK log level [7] freeswitch at internal> sofia USAGE: -------------------------------------------------------------------------------- sofia help sofia profile [[start|stop|restart|rescan] [reloadxml]|flush_inbound_reg [] [reboot]|[register|unregister] [|all]|killgw |[stun-auto-disable|stun-enabled] [true|false]]|siptrace [on|off] sofia status profile [ reg ] | [ pres ] sofia status gateway sofia loglevel [0-9] -------------------------------------------------------------------------------- freeswitch at internal> sofia loglevel all 9 Sofia log level for component [all] has been set to [9] freeswitch at internal> sofia profile internal siptrace on Enabled sip debugging on internal nua: nua_set_params: entering freeswitch at internal> nua((nil)): sent signal r_set_params nua((nil)): recv signal r_set_params nua: nua_stack_set_params: entering soa_set_params(static::0x9baf508, ...) called nua((nil)): event r_set_params 200 OK nua: nua_application_event: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9bcba58 from (udp/192.168.0.2:5070) has 1299 bytes, veclen = 1 recv 1299 bytes from udp/[192.168.0.2]:5060 at 23:06:35.605391: ------------------------------------------------------------------------ INVITE sip:0631000001 at 192.168.0.2:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Max-Forwards: 69 Contact: To: From: "0445674567";tag=50f70f41-co453-INS001 Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE Content-Type: application/sdp Date: Thu, 04 Jun 2009 23:06:20 GMT Supported: 100rel User-Agent: ENSR2.5.4 Content-Length: 288 P-Incoming-GW: Yes P-Incoming-GW: Yes X-Max-TIMER: 7200 X-Source_IP: 202.xx.xxx.xx P-Asserted-Identity: v=0 o=- 1358368577 1358368577 IN IP4 202.xx.xxx.xx s=ENSResip c=IN IP4 202..xxx.xx.xx t=0 0 m=audio 12580 RTP/AVP 18 8 0 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=sendrecv ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9bcba58 (1299 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received INVITE sip:0631000001 at 192.168.0.2:5060 SIP/2.0 (CSeq 45301) nta: INVITE (45301) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9ccc130) called soa_set_params(static::0x9c1d4e8, ...) called nta_leg_tcreate(0x9c41a68) soa_init_offer_answer(static::0x9c1d4e8) called soa_set_remote_sdp(static::0x9c1d4e8, (nil), 0x9c85df3, 288) called nua(0x9ccc130): adding session usage tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 698 bytes of 698 to udp/192.168.0.2:5060 tport_vsend returned 698 send 698 bytes to udp/[192.168.0.2]:5060 at 23:06:35.624664: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Record-Route: Record-Route: From: "0445674567";tag=50f70f41-co453-INS001 To: Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE User-Agent: Evolutiontel SIP Service Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (45301) nua(0x9ccc130): event i_invite 100 Trying nua(0x9ccc130): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x9c1d4e8, [0xb77b7bac], [0xb77b7ba8], [(nil)]) called nua(0x9ccc130): event i_state 100 Trying nua: nua_application_event: entering 2009-06-05 09:06:35 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/0445674567 at sip.evolutiontel.net [e57f4f54-7b45-48cc-98d3-8b594bdc19e8] nua: nua_handle_bind: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [received][100] 2009-06-05 09:06:35 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1358368577 1358368577 IN IP4 202.xx.xxx.xx s=ENSResip c=IN IP4 202..xxx.xx.xx t=0 0 m=audio 12580 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - 2009-06-05 09:06:35 [DEBUG] sofia.c:3182 sofia_handle_sip_i_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_NEW -> CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State INIT 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA INIT 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_INIT -> CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State INIT going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State ROUTING 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/0445674567 at sip.evolutiontel.net Standard ROUTING 2009-06-05 09:06:35 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 0445674567->0631000001 in context evolutiontel Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unloop] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->global] continue=true Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${network_addr}(192.168.0.2) =~ /^$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net ANTI-Action set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [global] ${numbering_plan}() =~ /^$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set_user(default@${domain_name}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_h_X-ZRTP-On}() =~ /^Y$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_secure_media}() =~ /^true$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_user_agent}(ENSR2.5.4) =~ /^PolycomSound(Point|Station)IP-S(S|P)IP_\d{3,4}-UA\/((3).(\d).(\d).(\d{4}))$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Absolute Condition [global] Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->vmain] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [vmain] destination_number(0631000001) =~ /^121/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->vmain1] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [vmain1] destination_number(0631000001) =~ /^123/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->vmain2] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [vmain2] destination_number(0631000001) =~ /^122/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->ivr_demo] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [ivr_demo] destination_number(0631000001) =~ /^5000$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(0631000001) =~ /^5900$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(0631000001) =~ /^5901$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(0631000001) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(0631000001) =~ /^parking$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(0631000001) =~ /callpark/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(0631000001) =~ /pickup/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->wait] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [wait] destination_number(0631000001) =~ /^wait$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->National_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [National_calls] destination_number(0631000001) =~ /^0(2|3|4|5|7|8|9)[0-9]{8}$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->Special_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [Special_calls] destination_number(0631000001) =~ /^1[3|8][0-9]+$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->International_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [International_calls] destination_number(0631000001) =~ /^0011[0-9]+$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->On-Net_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [On-Net_calls] destination_number(0631000001) =~ /^063[0-9]{7}$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(execute_on_answer=sched_hangup +${sip_h_x-max-timer} ALLOTED_TIMEOUT) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(sip_cid_type=pid) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(continue_on_fail=79) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(bypass_media=false) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(ringback=%(400,200,401,450);%(400,2200,400,450)) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action pre_answer() Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action export(sip_secure_media=true) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_ROUTING -> CS_EXECUTE 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State ROUTING going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_EXECUTE 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State EXECUTE 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA EXECUTE 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/0445674567 at sip.evolutiontel.net Standard EXECUTE EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(use_profile=nat) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [use_profile]=[nat] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set_user(default at 192.168.0.2) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(bypass_media=true) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(bypass_media=true) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net hash(insert/192.168.0.2-spymap/0445674567/e57f4f54-7b45-48cc-98d3-8b594bdc19e8) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/0445674567/0631000001) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/global/e57f4f54-7b45-48cc-98d3-8b594bdc19e8) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(execute_on_answer=sched_hangup +7200 ALLOTED_TIMEOUT) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [execute_on_answer]=[sched_hangup +7200 ALLOTED_TIMEOUT] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(sip_cid_type=pid) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [sip_cid_type]=[pid] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(continue_on_fail=79) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [continue_on_fail]=[79] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net bridge({sip_from_uri=sip:0445674567 at sip.evolutiontel.net}sofia/internal/0631000001 at 192.168.0.2^0312341234 at 202.xxx.xx.xx:5060) 2009-06-05 09:06:35 [DEBUG] switch_ivr_originate.c:1017 switch_ivr_originate() variable string 0 = [sip_from_uri=sip:0445674567 at sip.evolutiontel.net] 2009-06-05 09:06:35 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/0631000001 at 192.168.0.2 [d360e71f-64bd-4eb0-b99d-56af3c9e01f0] 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:2719 sofia_outgoing_channel() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State INIT 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/0631000001 at 192.168.0.2 SOFIA INIT nua: nh_create_handle: entering nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x9cf7ed0): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9cf7ed0) called soa_set_params(static::0x9d15ce0, ...) called soa_set_params(static::0x9d15ce0, ...) called soa_set_user_sdp(static::0x9d15ce0, (nil), 0x9cd16f7, -1) called soa_set_capability_sdp(static::0x9d15ce0, (nil), 0x9cd16f7, -1) called nua(0x9cf7ed0): adding session usage nta_leg_tcreate(0x9db94b0) soa_init_offer_answer(static::0x9d15ce0) called soa_generate_offer(static::0x9d15ce0, 0) called soa_static_offer_answer_action(0x9d15ce0, soa_generate_offer): called soa_static(0x9d15ce0, soa_generate_offer): generating local description soa_static(0x9d15ce0, soa_generate_offer): upgrade with local description soa_sdp_mode_set(0xb77b7ef4, (nil), ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.xxx.xx.xx (a local address) soa_static(0x9d15ce0, soa_generate_offer): storing local description soa_get_local_sdp(static::0x9d15ce0, [(nil)], [0xb77b7f7c], [0xb77b7f78]) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 1233 bytes of 1233 to udp/192.168.0.2:5060 tport_vsend returned 1233 send 1233 bytes to udp/[192.168.0.2]:5060 at 23:06:35.639526: ------------------------------------------------------------------------ INVITE sip:0631000001 at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5070;rport;branch=z9hG4bKZa7eS5j4r9a8F Max-Forwards: 68 From: "0445674567" ;tag=vQ8mytQ2mXNme To: Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Contact: User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 294 P-Incoming-GW: Yes P-Incoming-GW-1: Yes X-Max-TIMER: 7200 X-Source_IP: 202.xx.xxx.xx P-Asserted-Identity: "0445674567" v=0 o=- 7279717662006744601 8597592345307020033 IN IP4 202.xxx.xx.xx s=ENSResip c=IN IP4 202..xxx.xx.xx t=0 0 m=audio 12580 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - ------------------------------------------------------------------------ nta: sent INVITE (115960061) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9c63a58 for udp/192.168.0.2:5070 (already 0) nua(0x9cf7ed0): call state changed: init -> calling, sent offer soa_get_local_sdp(static::0x9d15ce0, [0xb77b7fa4], [0xb77b7fa0], [(nil)]) called nua(0x9cf7ed0): event i_state INVITE sent nua: nua_application_event: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c633c8 from (udp/192.168.0.2:5070) has 303 bytes, veclen = 1 recv 303 bytes from udp/[192.168.0.2]:5060 at 23:06:35.640683: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2:5070;rport=5070;branch=z9hG4bKZa7eS5j4r9a8F From: "0445674567" ;tag=vQ8mytQ2mXNme To: Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9c633c8 (303 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 100 Trying for INVITE (115960061) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 1.405 ms tport_release(0x9bb0e28): 0x9c63a58 by 0x9bcc770 with 0x9c633c8 (preliminary) nua(0x9cf7ed0): sent signal r_invite 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0631000001 at 192.168.0.2 entering state [calling][0] nua: nua_handle_magic: entering 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State INIT going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State ROUTING 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/0631000001 at 192.168.0.2 SOFIA ROUTING 2009-06-05 09:06:35 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State ROUTING going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_CONSUME_MEDIA 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State CONSUME_MEDIA 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State CONSUME_MEDIA going to sleep nta: timer not set tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c633c8 from (udp/192.168.0.2:5070) has 714 bytes, veclen = 1 recv 714 bytes from udp/[192.168.0.2]:5060 at 23:06:35.925933: ------------------------------------------------------------------------ SIP/2.0 180 Ringing To: ;tag=f936c99a6bd026a9i0 From: "0445674567" ;tag=vQ8mytQ2mXNme Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Via: SIP/2.0/UDP 192.168.0.2:5070;received=192.168.0.2;rport=5070;branch=z9hG4bKZa7eS5j4r9a8F Record-Route: Record-Route: Record-Route: Server: Linksys/SPA3000-3.1.20(GW) Remote-Party-ID: 0631000001 ;screen=yes;party=called Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9c633c8 (714 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 180 Ringing for INVITE (115960061) nta: 180 Ringing is going to a transaction tport_release(0x9bb0e28): 0x9c63a58 by 0x9bcc770 with 0x9c633c8 (preliminary) nua(0x9cf7ed0): event r_invite 180 Ringing nua(0x9cf7ed0): call state changed: calling -> proceeding nua(0x9cf7ed0): event i_state 180 Ringing nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0631000001 at 192.168.0.2 entering state [proceeding][180] 2009-06-05 09:06:35 [NOTICE] sofia.c:3103 sofia_handle_sip_i_state() Ring-Ready sofia/internal/0631000001 at 192.168.0.2! 2009-06-05 09:06:35 [DEBUG] sofia.c:3111 sofia_handle_sip_i_state() sofia/internal/0445674567 at sip.evolutiontel.net receive message [RINGING] nua: nua_respond: entering nua(0x9ccc130): sent signal r_respond 2009-06-05 09:06:35 [NOTICE] mod_sofia.c:1422 sofia_receive_message() Ring-Ready sofia/internal/0445674567 at sip.evolutiontel.net! 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering nua(0x9ccc130): recv signal r_respond 180 Ringing nua: nua_stack_set_params: entering soa_set_params(static::0x9c1d4e8, ...) called nua: nua_invite_server_respond: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1125 bytes of 1125 to udp/192.168.0.2:5060 tport_vsend returned 1125 send 1125 bytes to udp/[192.168.0.2]:5060 at 23:06:35.927247: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Record-Route: Record-Route: From: "0445674567";tag=50f70f41-co453-INS001 To: ;tag=UeFvvZ6yQmZ1j Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE Contact: User-Agent: Evolutiontel SIP Service Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 P-Asserted-Identity: "" <0631000001> ------------------------------------------------------------------------ nta: sent 180 Ringing for INVITE (45301) nta: timer set to 60000 ms nua(0x9ccc130): call state changed: received -> early nua(0x9ccc130): event i_state 180 Ringing nua: nua_application_event: entering 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [early][180] nua: nua_handle_magic: entering 2009-06-05 09:06:35 [DEBUG] switch_ivr_originate.c:1768 switch_ivr_originate() sofia/internal/0445674567 at sip.evolutiontel.net receive message [RINGING] 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:35 [NOTICE] switch_ivr_originate.c:1768 switch_ivr_originate() Ring Ready sofia/internal/0445674567 at sip.evolutiontel.net! tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9d93fd8 from (udp/192.168.0.2:5070) has 1218 bytes, veclen = 1 recv 1218 bytes from udp/[192.168.0.2]:5060 at 23:06:41.842064: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=f936c99a6bd026a9i0 From: "0445674567" ;tag=vQ8mytQ2mXNme Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Via: SIP/2.0/UDP 192.168.0.2:5070;received=192.168.0.2;rport=5070;branch=z9hG4bKZa7eS5j4r9a8F Record-Route: Record-Route: Record-Route: Contact: 0631000001 Server: Linksys/SPA3000-3.1.20(GW) Remote-Party-ID: 0631000001 ;screen=yes;party=called Content-Length: 306 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp P-Behind-NAT: Yes v=0 o=- 8503548 8503548 IN IP4 192.168.0.10 s=- c=IN IP4 60.xxx.xx.xxx t=0 0 m=audio 16582 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=direction:active a=oldmediaip:192.168.0.10 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9d93fd8 (1218 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for INVITE (115960061) nta: 200 OK is going to a transaction tport_release(0x9bb0e28): 0x9c63a58 by 0x9bcc770 with 0x9d93fd8 nta: timer shortened to 32000 ms soa_set_remote_sdp(static::0x9d15ce0, (nil), 0x9c62d18, 306) called soa_process_answer(static::0x9d15ce0) called soa_static_offer_answer_action(0x9d15ce0, soa_process_answer): called soa_sdp_mode_set(0x9cec6b0, 0x9db77b0, ""): called soa_static(0x9d15ce0, soa_process_answer): upgrade codecs with remote description soa_static(0x9d15ce0, soa_process_answer): storing local description soa_activate(static::0x9d15ce0, (nil)) called nua(0x9cf7ed0): INVITE: processed SDP answer in 200 OK nua(0x9cf7ed0): event r_invite 200 OK soa_activate(static::0x9d15ce0, (nil)) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 624 bytes of 624 to udp/192.168.0.2:5060 tport_vsend returned 624 send 624 bytes to udp/[192.168.0.2]:5060 at 23:06:41.842923: ------------------------------------------------------------------------ ACK sip:0631000001 at 60.xxx.xx.xxx:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5070;rport;branch=z9hG4bK0K07t037Nj1tB Route: Route: Route: Max-Forwards: 70 From: "0445674567" ;tag=vQ8mytQ2mXNme To: ;tag=f936c99a6bd026a9i0 Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ nta: sent ACK (115960061) to */192.168.0.2:5060 nua(0x9cf7ed0): call state changed: proceeding -> ready, received answer soa_get_remote_sdp(static::0x9d15ce0, [0xb77b7bfc], [0xb77b7bf8], [(nil)]) called soa_get_params(static::0x9d15ce0, ...) called nua(0x9cf7ed0): event i_state 200 OK nua(0x9cf7ed0): event i_active 200 Call active nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:41 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0631000001 at 192.168.0.2 entering state [ready][200] 2009-06-05 09:06:41 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 8503548 8503548 IN IP4 192.168.0.10 s=- c=IN IP4 60.xxx.xx.xxx t=0 0 m=audio 16582 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=direction:active a=oldmediaip:192.168.0.10 2009-06-05 09:06:41 [DEBUG] switch_channel.c:1875 switch_channel_perform_mark_answered() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:41 [DEBUG] switch_ivr_originate.c:1978 switch_ivr_originate() sofia/internal/0445674567 at sip.evolutiontel.net receive message [ANSWER] nua: nua_respond: entering nua(0x9ccc130): sent signal r_respond 2009-06-05 09:06:41 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:41 [NOTICE] switch_ivr_originate.c:1978 switch_ivr_originate() Channel [sofia/internal/0445674567 at sip.evolutiontel.net] has been answered 2009-06-05 09:06:41 [DEBUG] switch_channel.c:1911 switch_channel_perform_mark_answered() sofia/internal/0445674567 at sip.evolutiontel.net execute on answer: sched_hangup(+7200 ALLOTED_TIMEOUT) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net sched_hangup(+7200 ALLOTED_TIMEOUT) 2009-06-05 09:06:41 [DEBUG] switch_scheduler.c:214 switch_scheduler_add_task() Added task 7 switch_ivr_schedule_hangup (e57f4f54-7b45-48cc-98d3-8b594bdc19e8) to run at 1244164001 2009-06-05 09:06:41 [DEBUG] switch_ivr_originate.c:2024 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/0631000001 at 192.168.0.2] nua(0x9ccc130): recv signal r_respond 200 OK nua: nua_stack_set_params: entering soa_set_params(static::0x9c1d4e8, ...) called soa_set_user_sdp(static::0x9c1d4e8, (nil), 0x9dc2c08, -1) called soa_set_capability_sdp(static::0x9c1d4e8, (nil), 0x9dc2c08, -1) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x9c1d4e8) called soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called soa_static(0x9c1d4e8, soa_generate_answer): generating local description soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media soa_sdp_mode_set(0xb77b7fb4, 0x9ce6698, ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.xxx.xx.xx (a local address) soa_static(0x9c1d4e8, soa_generate_answer): storing local description soa_activate(static::0x9c1d4e8, (nil)) called soa_get_local_sdp(static::0x9c1d4e8, [(nil)], [0xb77b803c], [0xb77b8038]) called tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1414 bytes of 1414 to udp/192.168.0.2:5060 tport_vsend returned 1414 send 1414 bytes to udp/[192.168.0.2]:5060 at 23:06:41.847721: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Record-Route: Record-Route: From: "0445674567";tag=50f70f41-co453-INS001 To: ;tag=UeFvvZ6yQmZ1j Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE Contact: User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 256 P-Asserted-Identity: "" <0631000001> v=0 o=- 2963411278722346302 7975655717233217142 IN IP4 202.xxx.xx.xx s=- c=IN IP4 60.xxx.xx.xxx t=0 0 m=audio 0 RTP/AVP 96 100 101 a=rtpmap:96 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (45301) nta: timer shortened to 500 ms nua(0x9ccc130): call state changed: early -> completed, sent answer soa_get_local_sdp(static::0x9c1d4e8, [0xb77b8124], [0xb77b8120], [(nil)]) called soa_get_params(static::0x9c1d4e8, ...) called nua(0x9ccc130): event i_state 200 OK 2009-06-05 09:06:41 [DEBUG] switch_ivr_bridge.c:791 switch_ivr_signal_bridge() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:41 [DEBUG] switch_ivr_bridge.c:792 switch_ivr_signal_bridge() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State EXECUTE going to sleep 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HIBERNATE 2009-06-05 09:06:41 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/0445674567 at sip.evolutiontel.net Standard HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HIBERNATE going to sleep 2009-06-05 09:06:41 [NOTICE] sofia.c:3509 sofia_handle_sip_i_state() Channel [sofia/internal/0631000001 at 192.168.0.2] has been answered nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:41 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [completed][200] nua: nua_handle_magic: entering 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HIBERNATE 2009-06-05 09:06:41 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/0631000001 at 192.168.0.2 SOFIA HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/0631000001 at 192.168.0.2 Standard HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HIBERNATE going to sleep tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9cbc8b8 from (udp/192.168.0.2:5070) has 475 bytes, veclen = 1 recv 475 bytes from udp/[192.168.0.2]:5060 at 23:06:42.052636: ------------------------------------------------------------------------ ACK sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.2 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-1 Max-Forwards: 69 To: ;tag=UeFvvZ6yQmZ1j From: "0445674567";tag=50f70f41-co453-INS001 Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 ACK User-Agent: ENSR2.5.4 Content-Length: 0 P-hint: rr-enforced ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9cbc8b8 (475 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received ACK sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 45301) nta: ACK (45301) is going to INVITE (45301) nua: process_ack_or_cancel: entering soa_clear_remote_sdp(static::0x9c1d4e8) called nua(0x9ccc130): event i_ack 200 OK nua(0x9ccc130): call state changed: completed -> ready nua(0x9ccc130): event i_state 200 OK nua(0x9ccc130): event i_active 200 Call active nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:42 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [ready][200] nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9cbc8b8 from (udp/192.168.0.2:5070) has 475 bytes, veclen = 1 recv 475 bytes from udp/[192.168.0.2]:5060 at 23:06:42.088227: ------------------------------------------------------------------------ BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK62ec.b694c4f3.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f6-0 Max-Forwards: 69 To: ;tag=UeFvvZ6yQmZ1j From: "0445674567";tag=50f70f41-co453-INS001 Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45302 BYE User-Agent: ENSR2.5.4 Content-Length: 0 P-hint: rr-enforced ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9cbc8b8 (475 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 45302) nta: canonizing sip:mod_sofia at 192.168.0.2:5070 with contact nta: BYE (45302) going to existing leg nua: nua_stack_process_request: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 588 bytes of 588 to udp/192.168.0.2:5060 tport_vsend returned 588 send 588 bytes to udp/[192.168.0.2]:5060 at 23:06:42.088632: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK62ec.b694c4f3.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f6-0 From: "0445674567";tag=50f70f41-co453-INS001 To: ;tag=UeFvvZ6yQmZ1j Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45302 BYE User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for BYE (45302) nua(0x9ccc130): event i_bye 200 Session Terminated nua(0x9ccc130): removing session usage nua(0x9ccc130): call state changed: ready -> terminated nua(0x9ccc130): event i_state 200 Session Terminated nua(0x9ccc130): event i_terminated 200 Session Terminated soa_destroy(static::0x9c1d4e8) called nta_leg_destroy(0x9c41a68) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:42 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [terminated][200] 2009-06-05 09:06:42 [NOTICE] sofia.c:3599 sofia_handle_sip_i_state() Hangup sofia/internal/0445674567 at sip.evolutiontel.net [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-05 09:06:42 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [KILL] 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua(0x9ccc130): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua: nua_application_event: entering nua(0x9ccc130): event i_terminated dropped 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_HANGUP 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HANGUP 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/0445674567 at sip.evolutiontel.net Overriding SIP cause 480 with 200 from the other leg 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/0445674567 at sip.evolutiontel.net hanging up, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [NOTICE] switch_ivr_bridge.c:712 signal_bridge_on_hangup() Hangup sofia/internal/0631000001 at 192.168.0.2 [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-05 09:06:42 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/0631000001 at 192.168.0.2 [KILL] 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/0445674567 at sip.evolutiontel.net Standard HANGUP, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HANGUP going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_HANGUP -> CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State REPORTING nua(0x9ccc130): recv signal r_destroy nta_leg_destroy((nil)) 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_HANGUP 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HANGUP 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/0631000001 at 192.168.0.2 Overriding SIP cause 480 with 200 from the other leg 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/0631000001 at 192.168.0.2 hanging up, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:394 sofia_on_hangup() Sending BYE to sofia/internal/0631000001 at 192.168.0.2 nua: nua_bye: entering nua(0x9cf7ed0): sent signal r_bye 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/0631000001 at 192.168.0.2 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HANGUP going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0631000001 at 192.168.0.2) State REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/0631000001 at 192.168.0.2 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0631000001 at 192.168.0.2) State REPORTING going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 10 (sofia/internal/0631000001 at 192.168.0.2) Locked, Waiting on external entities 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 10 (sofia/internal/0631000001 at 192.168.0.2) Ended 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/0631000001 at 192.168.0.2 [CS_DESTROY] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0631000001 at 192.168.0.2) State DESTROY 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/0631000001 at 192.168.0.2 SOFIA DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/0631000001 at 192.168.0.2 Standard DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0631000001 at 192.168.0.2) State DESTROY going to sleep nua(0x9cf7ed0): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x9d15ce0, ...) called soa_terminate(static::0x9d15ce0) called soa_init_offer_answer(static::0x9d15ce0) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 874 bytes of 874 to udp/192.168.0.2:5060 tport_vsend returned 874 send 874 bytes to udp/[192.168.0.2]:5060 at 23:06:42.094504: ------------------------------------------------------------------------ BYE sip:0631000001 at 60.xxx.xx.xxx:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5070;rport;branch=z9hG4bK1vS0vUmBKUQDQ Route: Route: Route: Max-Forwards: 70 From: "0445674567" ;tag=vQ8mytQ2mXNme To: ;tag=f936c99a6bd026a9i0 Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960062 BYE Contact: User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ nta: sent BYE (115960062) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9d15e08 for udp/192.168.0.2:5070 (already 0) 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/0445674567 at sip.evolutiontel.net Standard REPORTING, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State REPORTING going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_REPORTING -> CS_DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 9 (sofia/internal/0445674567 at sip.evolutiontel.net) Locked, Waiting on external entities 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 9 (sofia/internal/0445674567 at sip.evolutiontel.net) Ended 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/0445674567 at sip.evolutiontel.net [CS_DESTROY] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State DESTROY 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/0445674567 at sip.evolutiontel.net Standard DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State DESTROY going to sleep tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c41b38 from (udp/192.168.0.2:5070) has 450 bytes, veclen = 1 recv 450 bytes from udp/[192.168.0.2]:5060 at 23:06:42.210621: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=f936c99a6bd026a9i0 From: "0445674567" ;tag=vQ8mytQ2mXNme Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960062 BYE Via: SIP/2.0/UDP 192.168.0.2:5070;received=192.168.0.2;rport=5070;branch=z9hG4bK1vS0vUmBKUQDQ Server: Linksys/SPA3000-3.1.20(GW) P-RTP-Stat: PS=12,OS=240,PR=0,OR=0,PL=0,JI=0,LA=0,DU=0,EN=G729a,DE=G711u Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9c41b38 (450 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for BYE (115960062) nta: 200 OK is going to a transaction nta_outgoing: RTT is 116.446 ms tport_release(0x9bb0e28): 0x9d15e08 by 0x9bcdcb0 with 0x9c41b38 nua(0x9cf7ed0): event r_bye 200 OK nua(0x9cf7ed0): call state changed: terminating -> terminated nua(0x9cf7ed0): event i_state 200 to BYE nua(0x9cf7ed0): event i_terminated 200 to BYE nua(0x9cf7ed0): removing session usage soa_destroy(static::0x9d15ce0) called nta_leg_destroy(0x9db94b0) nua: terminated session 0x9cf7ed0 nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x9cf7ed0): sent signal r_destroy nua: nua_application_event: entering nua(0x9cf7ed0): event i_terminated dropped nua(0x9cf7ed0): recv signal r_destroy nta_leg_destroy((nil)) nta: timer set next to 4704 ms 2009-06-05 09:06:42 [DEBUG] switch_scheduler.c:138 task_thread_loop() Deleting task 7 switch_ivr_schedule_hangup (e57f4f54-7b45-48cc-98d3-8b594bdc19e8) nta: timer I fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0xb77b81cc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 157 ms nta: timer K fired, terminate BYE (115960062) outgoing_reclaim_all((nil), (nil), 0xb77b81c8) nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/2 term, 1/3 free nta: timer set next to 26628 ms /exit [root at sip01 bin]# From jim at evolutiontel.net Thu Jun 4 16:18:36 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 09:18:36 +1000 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: <079f01c9e535$c1fed2e0$45fc78a0$@com> References: <079f01c9e535$c1fed2e0$45fc78a0$@com> Message-ID: Not that this helps your question directly. Using a putty terminal from windows allows the data to be copied correctly from console sessions. On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: > I want to copy the results of a siptrace captured on the fs console to a > file. The console is running on a Gnome terminal. I highlight the text I > want to copy in the fs console, open a vi session in insert mode, and paste > the text. However, the text is not pasted as I copied it ? it is missing > characters/lines. > > > > I know I am doing something wrong. Is there another way to save siptraces to > a file? Redirection doesn?t work. > > > > sofia profile internal siptrace on is the command I use. > > > > Thanks Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From brian at freeswitch.org Thu Jun 4 16:28:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 18:28:57 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> Message-ID: <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> Port 0 indicates a rejection. Something is sending a 200 OK with 0 and g729a on codec 96. Have you modified the freeswitch code ? /b On Jun 4, 2009, at 6:15 PM, Jim Burke wrote: > Hi Anthony, > > Traces as requested. Let me know if you want a jira opened or any > further data. > > Regards, > Jim Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0e46b828/attachment.html From msc at freeswitch.org Thu Jun 4 16:29:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 16:29:18 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> Message-ID: <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood wrote: > Okay, leave this with me - I'll bring this together and do what's required. > I'll see if I can squeeze it in ahead of schedule, but don't count on it. > Also expect pleas for help at some stage, and I'll need a UI developer to > create an awesome interface. > > You'll all be hearing from me shortly! > You're hired!!! :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/5b561c5c/attachment.html From matthew.lockwood at gmail.com Thu Jun 4 16:36:26 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 16:36:26 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> Message-ID: <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> Awesome :-D I need some thinking time to come up with a strategy, but I'll set up a project site soon so we can all collaborate on this. M On Thu, Jun 4, 2009 at 4:29 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> Okay, leave this with me - I'll bring this together and do what's >> required. I'll see if I can squeeze it in ahead of schedule, but don't count >> on it. Also expect pleas for help at some stage, and I'll need a UI >> developer to create an awesome interface. >> >> You'll all be hearing from me shortly! >> > > You're hired!!! :) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/f2d9ecce/attachment.html From msc at freeswitch.org Thu Jun 4 16:47:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 16:47:20 -0700 Subject: [Freeswitch-users] Getting Ready To Welcome The Newest Little FreeSWITCH Community Member! Message-ID: <87f2f3b90906041647w727e04c0je357c2e2bad32de7@mail.gmail.com> Hey everyone, Raymond Chandler (aka Intralanman) and his wife are expecting a baby boy! Check out the story and how you can help welcome our newest FreeSWITCH community member: http://www.freeswitch.org/node/189 -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/9d75f6df/attachment.html From jim at evolutiontel.net Thu Jun 4 16:49:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 09:49:39 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> Message-ID: Using FreeSWITCH Version 1.0.trunk (13523) and have not modified the code. Yes, exactly this is what causes the originator to release the call. As you will see in the traces, the 200OK is good on the way into FS, but looks to be trampled on the way out :( Regards, On Fri, Jun 5, 2009 at 9:28 AM, Brian West wrote: > Port 0 indicates a rejection. ?Something is sending a 200 OK with 0 and > g729a on codec 96. ?Have you modified the freeswitch code ? > /b > On Jun 4, 2009, at 6:15 PM, Jim Burke wrote: > > Hi Anthony, > > Traces as requested. ?Let me know if you want a jira opened or any further > data. > > Regards, > Jim > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From brian at freeswitch.org Thu Jun 4 16:51:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 18:51:13 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> Message-ID: <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> But that SDP is not generated by FreeSWITCH... are you using proxy media or something? maybe bypass? /b On Jun 4, 2009, at 6:49 PM, Jim Burke wrote: > Yes, exactly this is what causes the originator to release the call. > As you will see in the traces, the 200OK is good on the way into FS, > but looks to be trampled on the way out :( > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/371a927e/attachment.html From d at unwire.it Thu Jun 4 16:58:23 2009 From: d at unwire.it (Darin Weeks) Date: Thu, 4 Jun 2009 16:58:23 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> Message-ID: <989132e70906041658xba4f660u91ae3793f581f327@mail.gmail.com> I'm game to help out. I have "product" experience with UI and stuff... (but please don't think that I'm a business wonk). As a new freeswitch user it might be slightly harder to work with me rather than someone who knows the nuts and bolts of the system, but on the other hand, my n00b status might yield a more intuitive UI because it'll have to make sense to me in order to design it. Please keep me in the loop or add me to any emails on the topic and I'll see how I can join in. On Thu, Jun 4, 2009 at 4:36 PM, Matthew Lockwood wrote: > Awesome :-D > > I need some thinking time to come up with a strategy, but I'll set up a > project site soon so we can all collaborate on this. > > M > > On Thu, Jun 4, 2009 at 4:29 PM, Michael Collins wrote: > >> >> >> On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood < >> matthew.lockwood at gmail.com> wrote: >> >>> Okay, leave this with me - I'll bring this together and do what's >>> required. I'll see if I can squeeze it in ahead of schedule, but don't count >>> on it. Also expect pleas for help at some stage, and I'll need a UI >>> developer to create an awesome interface. >>> >>> You'll all be hearing from me shortly! >>> >> >> You're hired!!! :) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/50993fcd/attachment-0001.html From matthew.lockwood at gmail.com Thu Jun 4 17:03:26 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 17:03:26 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <989132e70906041658xba4f660u91ae3793f581f327@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> <989132e70906041658xba4f660u91ae3793f581f327@mail.gmail.com> Message-ID: <415541b10906041703s6f1e6a1dwaf0930023b401679@mail.gmail.com> Okay great. What I'll be doing is creating the back-end. When the project kicks off, it would be far better for someone else to design a pretty and useable UI on top of that. You'd not really need to know much about the inner workings of FS. I'll probably just end up knocking up a fugly unstyled HTML interface for testing purposes, so you could just take it from there. I'll be the business wonk! I'll be setting up a project site on my company's intranet soon. On Thu, Jun 4, 2009 at 4:58 PM, Darin Weeks wrote: > I'm game to help out. I have "product" experience with UI and stuff... > (but please don't think that I'm a business wonk). As a new freeswitch user > it might be slightly harder to work with me rather than someone who knows > the nuts and bolts of the system, but on the other hand, my n00b status > might yield a more intuitive UI because it'll have to make sense to me in > order to design it. > > Please keep me in the loop or add me to any emails on the topic and I'll > see how I can join in. > > > On Thu, Jun 4, 2009 at 4:36 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> Awesome :-D >> >> I need some thinking time to come up with a strategy, but I'll set up a >> project site soon so we can all collaborate on this. >> >> M >> >> On Thu, Jun 4, 2009 at 4:29 PM, Michael Collins wrote: >> >>> >>> >>> On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood < >>> matthew.lockwood at gmail.com> wrote: >>> >>>> Okay, leave this with me - I'll bring this together and do what's >>>> required. I'll see if I can squeeze it in ahead of schedule, but don't count >>>> on it. Also expect pleas for help at some stage, and I'll need a UI >>>> developer to create an awesome interface. >>>> >>>> You'll all be hearing from me shortly! >>>> >>> >>> You're hired!!! :) >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/e43c4748/attachment.html From jim at evolutiontel.net Thu Jun 4 17:07:44 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 10:07:44 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> Message-ID: bypass_media is set, and proxy_media is not set. I agree, FS should not be touching the SDP for calls in bypass_media mode. Interesting, when you look in the file when FS reports the Remote SDP it still looks ok, then a little further down you can see it send the 200OK out to the originator and that SDP in-correctly reports the media port. The other interesting point is a=rtpmap:96 G729a/8000. 96 is not the correct rtpmap for G729 and it is not mentioned on the incoming 200Ok to FS Regards, On Fri, Jun 5, 2009 at 9:51 AM, Brian West wrote: > But that SDP is not generated by FreeSWITCH... are you using proxy media or > something? ?maybe bypass? > /b > On Jun 4, 2009, at 6:49 PM, Jim Burke wrote: > > Yes, exactly this is what causes the originator to release the call. > As you will see in the traces, the 200OK is good on the way into FS, > but looks to be trampled on the way out :( > > Regards, > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From jim at evolutiontel.net Thu Jun 4 17:23:36 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 10:23:36 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> Message-ID: Hey Brian >From your comments above this appears to be the code that does the damage. I guess now the question is why?? soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called soa_static(0x9c1d4e8, soa_generate_answer): generating local description soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media Regards, On Fri, Jun 5, 2009 at 10:07 AM, Jim Burke wrote: > bypass_media is set, and proxy_media is not set. > > I agree, FS should not be touching the SDP for calls in bypass_media > mode. ?Interesting, when you look in the file when FS reports the > Remote SDP it still looks ok, then a little further down you can see > it send the 200OK out to the originator and that SDP in-correctly > reports the media port. > > The other interesting point is a=rtpmap:96 G729a/8000. ?96 is not the > correct rtpmap for G729 and it is not mentioned on the incoming 200Ok > to FS > > Regards, > > > On Fri, Jun 5, 2009 at 9:51 AM, Brian West wrote: >> But that SDP is not generated by FreeSWITCH... are you using proxy media or >> something? ?maybe bypass? >> /b >> On Jun 4, 2009, at 6:49 PM, Jim Burke wrote: >> >> Yes, exactly this is what causes the originator to release the call. >> As you will see in the traces, the 200OK is good on the way into FS, >> but looks to be trampled on the way out :( >> >> Regards, >> >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! ?http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From brian at freeswitch.org Thu Jun 4 17:34:17 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 19:34:17 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> Message-ID: <53828315-70FE-4A7E-A0B0-3608E16FCF56@freeswitch.org> Try SVN trunk I cna tell you're using older code! ;) /b On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: > Hey Brian > >> From your comments above this appears to be the code that does the > damage. I guess now the question is why?? > > soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called > soa_static(0x9c1d4e8, soa_generate_answer): generating local > description > soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote > description > soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/e7ba7835/attachment.html From jim at evolutiontel.net Thu Jun 4 18:00:28 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 11:00:28 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <53828315-70FE-4A7E-A0B0-3608E16FCF56@freeswitch.org> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> <53828315-70FE-4A7E-A0B0-3608E16FCF56@freeswitch.org> Message-ID: Hmmm...no luck with the SVN trunk FreeSWITCH Version 1.0.trunk (13624) On Fri, Jun 5, 2009 at 10:34 AM, Brian West wrote: > Try SVN trunk I cna tell you're using older code! ?;) > /b > On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: > > Hey Brian > > From your comments above this appears to be the code that does the > > damage. ?I guess now the question is why?? > > soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called > soa_static(0x9c1d4e8, soa_generate_answer): generating local description > soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description > soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media > > Regards, > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From dujinfang at gmail.com Thu Jun 4 18:38:46 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 09:38:46 +0800 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: <079f01c9e535$c1fed2e0$45fc78a0$@com> References: <079f01c9e535$c1fed2e0$45fc78a0$@com> Message-ID: <1E2BD79E-D1B8-41D1-8560-791F25A68E5A@gmail.com> try this: copy the text open a new terminal cat > siptrace.txt paste the text press [Enter] press [Ctrl] + d On Jun 5, 2009, at 12:59 AM, Lars Zeb wrote: > I want to copy the results of a siptrace captured on the fs console > to a file. The console is running on a Gnome terminal. I highlight > the text I want to copy in the fs console, open a vi session in > insert mode, and paste the text. However, the text is not pasted as > I copied it ? it is missing characters/lines. > > I know I am doing something wrong. Is there another way to save > siptraces to a file? Redirection doesn?t work. > > sofia profile internal siptrace on is the command I use. > > Thanks Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/9117d3c8/attachment-0001.html From larclap at yahoo.com Thu Jun 4 18:38:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 4 Jun 2009 18:38:45 -0700 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: References: <079f01c9e535$c1fed2e0$45fc78a0$@com> Message-ID: <08bc01c9e57e$5ef60030$1ce20090$@com> Does this mean that I must start fs from the putty terminal, or can I attach to an already running instance via putty? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim Burke Sent: Thursday, June 04, 2009 4:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing lines copying data from console to vi Not that this helps your question directly. Using a putty terminal from windows allows the data to be copied correctly from console sessions. On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: > I want to copy the results of a siptrace captured on the fs console to a > file. The console is running on a Gnome terminal. I highlight the text I > want to copy in the fs console, open a vi session in insert mode, and paste > the text. However, the text is not pasted as I copied it - it is missing > characters/lines. > > > > I know I am doing something wrong. Is there another way to save siptraces to > a file? Redirection doesn't work. > > > > sofia profile internal siptrace on is the command I use. > > > > Thanks Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From intralanman at freeswitch.org Thu Jun 4 18:40:12 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 04 Jun 2009 21:40:12 -0400 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041539r47bdf2c6m1df4671ad4f67342@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <87f2f3b90906041539r47bdf2c6m1df4671ad4f67342@mail.gmail.com> Message-ID: <4A28777C.1000609@freeswitch.org> we created a #freeswitch-gui channel for talks such as these... I'm the only one idling in it most of the time, though :-( -Ray Michael Collins wrote: > > On Thu, Jun 4, 2009 at 2:39 PM, Matthew Lockwood > > wrote: > > I'm about to start creating one. I think FS needs a UI comparable > to FreePBX or similar. It's the next project on my list. It would > certainly be a boost to the FS project and user acceptance. > > > Be sure to check with Bougyman on IRC. He is planning on releasing an > open-source, MIT-licensed FS-GUI w/ underlying framework. Last I heard > he said week of June 15. Note: he said that it will require Ruby plus > Rack and PostgreSQL. > > -MC > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/8c050b6a/attachment.html From edpimentl at gmail.com Thu Jun 4 18:47:54 2009 From: edpimentl at gmail.com (EdPimentl) Date: Thu, 4 Jun 2009 21:47:54 -0400 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> Message-ID: <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> I would like to see it, and you would get many more volunteers if it would jquery/mootools UI with DJANGO backend. Or a MOZILLA XUL Or a DOJO APP Or a FLEX/AIR That would be different and unique.... -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/aa623596/attachment.html From jim at evolutiontel.net Thu Jun 4 18:51:32 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 11:51:32 +1000 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: <08bc01c9e57e$5ef60030$1ce20090$@com> References: <079f01c9e535$c1fed2e0$45fc78a0$@com> <08bc01c9e57e$5ef60030$1ce20090$@com> Message-ID: After logging into my linux box using putty. I then change directory the the ~/freeswitch/bin directory and run ./fs_cli You can, but you don't need to start FS from the putty terminal. We run ours as a background process. On Fri, Jun 5, 2009 at 11:38 AM, Lars Zeb wrote: > Does this mean that I must start fs from the putty terminal, or can I attach > to an already running instance via putty? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim > Burke > Sent: Thursday, June 04, 2009 4:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Missing lines copying data from console to > vi > > Not that this helps your question directly. ?Using a putty terminal > from windows allows the data to be copied correctly from console > sessions. > > > > On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: >> I want to copy the results of a siptrace captured on the fs console to a >> file. The console is running on a Gnome terminal. I highlight the text I >> want to copy in the fs console, open a vi session in insert mode, and > paste >> the text. However, the text is not pasted as I copied it - it is missing >> characters/lines. >> >> >> >> I know I am doing something wrong. Is there another way to save siptraces > to >> a file? Redirection doesn't work. >> >> >> >> sofia profile internal siptrace on is the command I use. >> >> >> >> Thanks Lars >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From matthew.lockwood at gmail.com Thu Jun 4 18:52:51 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 18:52:51 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> Message-ID: <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. On Thu, Jun 4, 2009 at 6:47 PM, EdPimentl wrote: > I would like to see it, and you would get many more volunteers if it would > jquery/mootools UI with DJANGO backend. > > Or a MOZILLA XUL > > Or a DOJO APP > > Or a FLEX/AIR > > That would be different and unique.... > -E > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/dfe9dc94/attachment.html From dujinfang at gmail.com Thu Jun 4 18:59:10 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 09:59:10 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <200906041628066407689@gmail.com> References: <200906041628066407689@gmail.com> Message-ID: <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then. On Jun 4, 2009, at 4:28 PM, god.nirvana wrote: > hi all > i am new to freeswitch. > there are some busy tone detect issues,i hope someone could help me. > i installed freeswitch from trunk,openzap,zaptel.... > but i found some busy tone isuues > > my tones.conf: > [us] > generate-dial => v=-7;%(1000,0,350,440) > detect-dial => 350,440 > generate-ring => v=-7;%(2000,4000,440,480) > detect-ring => 440,480 > generate-busy => v=-7;%(500,500,450,340) > detect-busy =>450,340 > generate-attn => v=0;%(100,100,1400,2060,2450,2600) > detect-attn => 1400,2060,2450,2600 > generate-callwaiting-sas => v=0;%(300,0,440) > detect-callwaiting-sas => 440 > generate-callwaiting-cas => v=0;%(80,0,2750,2130) > detect-callwaiting-cas => 2750,2130 > detect-fail1 => 913.8 > detect-fail2 => 1370.6 > detect-fail3 => 776.7 > > > openzap.conf.xml : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > when i call the pstn phone from a ip phone,if the pstn call hangup > first,the ip phone will hear the busy tone,but the system does not > handle the busytone ,the channel does not erase. so i have to add > in the dialplan.and it works.the channel > erased. > > but in the conference case,pstn phone call in,hangup. all > participants hear the tone,"do ~,do~~".freeswitch doest handle it. > so i change the conference dialplan. > > > > > > > > > > restart freeswitch,try again,freeswitch not handle the hangup tone > still,all participants hear the tone. > how to solve it?could some one help me ??? > thx! > BR > M > .Q > 2009-06-04 > god.nirvana > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/dd27b14e/attachment-0001.html From brad.tuan at gmail.com Thu Jun 4 19:43:40 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Fri, 5 Jun 2009 10:43:40 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: Yes I have tried it,but useless, when 1001 and 1002 are talking to each other , then 1001 want to transfer 1002 to 1003, so 1001 press *1 1003, but nothing happen....... 2009/6/4 dujinfang > yes. Did you ever tried that? > > On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: > > I mean when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003. > 2009/6/4 seven > >> the default config allows 1002 press *1 and 1003 to do blind transfer, >> also you may interest the att_xfer, see dp_tools on wiki. >> >> >> On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >> >> If i don't want to use softphone function to transfer the call ,how to do >> it?? >> >> 2009/6/4 Brian West >> >>> Depends.. Press the transfer key on your phone is how I would do it.. >>> what kind of phone do you have? >>> /b >>> >>> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >>> >>> When User(1001) calling with User(1002) , >>> >>> how to transfer User(1002) to User(1003)?? >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/c6147ea8/attachment.html From brian at freeswitch.org Thu Jun 4 19:50:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 21:50:13 -0500 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: Please show us your example of using att_xfr /b On Jun 4, 2009, at 9:43 PM, Brad Tuan wrote: > Yes I have tried it,but useless, > > when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003, > > so 1001 press *1 1003, > > but nothing happen....... Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/c1473300/attachment.html From dujinfang at gmail.com Thu Jun 4 20:09:12 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 11:09:12 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: guess if you press *1 on 1002 you can transfer 1001 to 1003 if you want to press *1 on 1001, find Local_extension in dianplan/ default.xml where data = "0 b ... means it's only works on the b-leg try change to a (for a-leg) or ab for both leg. check bind_meta_app for detail on wiki, I bet you never tried the att_xfer feature. On Jun 5, 2009, at 10:43 AM, Brad Tuan wrote: > Yes I have tried it,but useless, > > when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003, > > so 1001 press *1 1003, > > but nothing happen....... > 2009/6/4 dujinfang > yes. Did you ever tried that? > > On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: >> I mean when 1001 and 1002 are talking to each other , >> >> then 1001 want to transfer 1002 to 1003. >> 2009/6/4 seven >> the default config allows 1002 press *1 and 1003 to do blind >> transfer, also you may interest the att_xfer, see dp_tools on wiki. >> >> >> On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >>> If i don't want to use softphone function to transfer the >>> call ,how to do it?? >>> >>> 2009/6/4 Brian West >>> Depends.. Press the transfer key on your phone is how I would do >>> it.. what kind of phone do you have? >>> >>> /b >>> >>> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >>> >>>> When User(1001) calling with User(1002) , >>>> >>>> how to transfer User(1002) to User(1003)?? >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/d62d6649/attachment-0001.html From brad.tuan at gmail.com Thu Jun 4 21:37:27 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Fri, 5 Jun 2009 12:37:27 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: Yes I got it is the call leg setting problem,thanks 2009/6/5 seven > guess if you press *1 on 1002 you can transfer 1001 to 1003 > if you want to press *1 on 1001, find Local_extension in > dianplan/default.xml > > where data = "0 b ... means it's only works on the b-leg > > try change to a (for a-leg) or ab for both leg. > > > > check bind_meta_app for detail on wiki, I bet you never tried the att_xfer > feature. > > On Jun 5, 2009, at 10:43 AM, Brad Tuan wrote: > > Yes I have tried it,but useless, > > when 1001 and 1002 are talking to each other , > then 1001 want to transfer 1002 to 1003, > > so 1001 press *1 1003, > > but nothing happen....... > 2009/6/4 dujinfang > >> yes. Did you ever tried that? >> >> On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: >> >> I mean when 1001 and 1002 are talking to each other , >> >> then 1001 want to transfer 1002 to 1003. >> 2009/6/4 seven >> >>> the default config allows 1002 press *1 and 1003 to do blind transfer, >>> also you may interest the att_xfer, see dp_tools on wiki. >>> >>> >>> On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >>> >>> If i don't want to use softphone function to transfer the call ,how to do >>> it?? >>> >>> 2009/6/4 Brian West >>> >>>> Depends.. Press the transfer key on your phone is how I would do it.. >>>> what kind of phone do you have? >>>> /b >>>> >>>> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >>>> >>>> When User(1001) calling with User(1002) , >>>> >>>> how to transfer User(1002) to User(1003)?? >>>> >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/573039a1/attachment.html From msc at freeswitch.org Thu Jun 4 22:14:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 22:14:50 -0700 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> Message-ID: <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > I'm using openzap analog with tone_detect, it works(conference not tested). > however, according to the asterisk book, Kewlstart can detect the busy tone > and disconnect the circuit. does anyone knows how to configure kewlstart > with freeswitch/openzap? guess we don't need tone_detect then. > Dujinfang, Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.) Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/417cbbd8/attachment.html From stevecrozz at gmail.com Thu Jun 4 22:16:40 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 4 Jun 2009 22:16:40 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> Message-ID: <11990ade0906042216p13f8b05au778cb12b379ef98a@mail.gmail.com> In case anyone was wondering, I could lend a hand if django/python, ruby on rails, or php on were involved. --Stephen On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood wrote: > That would involve me learning a totally new framework. It'll not the > hardest code I'll ever write by far, so I'm okay coding it up on my own. > However, I definitely need a lot of help from fabulous designers to actually > make the interface pretty and useable. Plus, I'm only one person and will > need a lot of feedback to create something that rocks - everybody has a > different use case and I can't foresee how everybody will use it, so that > kind of feedback will go into re-engineering it. > > On Thu, Jun 4, 2009 at 6:47 PM, EdPimentl wrote: > >> I would like to see it, and you would get many more volunteers if it would >> jquery/mootools UI with DJANGO backend. >> >> Or a MOZILLA XUL >> >> Or a DOJO APP >> >> Or a FLEX/AIR >> >> That would be different and unique.... >> -E >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0a99f98e/attachment.html From msc at freeswitch.org Thu Jun 4 22:36:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 22:36:05 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> Message-ID: <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood wrote: > That would involve me learning a totally new framework. It'll not the > hardest code I'll ever write by far, so I'm okay coding it up on my own. > However, I definitely need a lot of help from fabulous designers to actually > make the interface pretty and useable. Plus, I'm only one person and will > need a lot of feedback to create something that rocks - everybody has a > different use case and I can't foresee how everybody will use it, so that > kind of feedback will go into re-engineering it. > If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited opinions about all sorts of things. :) All that being said, I say go for it. Find what works for you and see what happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure that we could even start a mailing list for GUI development. Enjoy! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0033a8f9/attachment-0001.html From jim at evolutiontel.net Fri Jun 5 01:06:41 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 18:06:41 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. Message-ID: <3wmhHdntwTK5.tvBsy3CM@smtp.gmail.com> should open a jira on this? Did you guys need any more info? - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. From: Brian West Date: 05/06/2009 00:35 Try SVN trunk I cna tell you're using older code! ;) /b On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: > Hey Brian > >> From your comments above this appears to be the code that does the > damage. I guess now the question is why?? > > soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called > soa_static(0x9c1d4e8, soa_generate_answer): generating local > description > soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote > description > soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Fri Jun 5 01:26:11 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 16:26:11 +0800 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> Message-ID: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood > wrote: > That would involve me learning a totally new framework. It'll not > the hardest code I'll ever write by far, so I'm okay coding it up on > my own. However, I definitely need a lot of help from fabulous > designers to actually make the interface pretty and useable. Plus, > I'm only one person and will need a lot of feedback to create > something that rocks - everybody has a different use case and I > can't foresee how everybody will use it, so that kind of feedback > will go into re-engineering it. > > If you guys are serious about this then I would like to make a few > suggestions that might be obvious but for the sake of the project > we'll make them explicitly obvious. > > First, before deciding what framework to use, it would be good to > hold some discussions about what the GUI actually needs to do: > What are the design goals? > Will it be just for setting up extensions and the dialplan? Or will > it go much farther than that? > Will you be using mod_xml_curl for everything? If so, what > database(s) will you support? > Are you going to have extra goodies like an IVR builder? > A 'visual voicemail' page? > A user portal? > Management interface to 'spy' on users? > A CDR/call accounting system? > FIFO and/or ACD queue management? > MOH and sound files management? and conference management > > > It's okay to start small and build your way out, but you need to > know before you start building what the grand scheme will be. The > larger the goals of the project, the narrower your choices for a > framework that can do it all. The simple fact of the matter is that > if you want to use a MVC web framework then you have a somewhat > limited number of choices. You need a MVC WF that fits your needs, > which means it needs to be at least somewhat flexible. If you want a > pretty GUI then you need to decide if you want a rich Internet > application (RIA) front end like AIR, or do you want something along > the lines of XHTML/CSS/JS and use a platform like Dojo which gives > you cross-browser widgets and tools. All of this on top of the fact > that if you want volunteers to assist you will need to pick > something that people either know or can learn quickly. > > Oh, and be prepared for people to give you unsolicited opinions > about all sorts of things. :) > > All that being said, I say go for it. Find what works for you and > see what happens. Be sure to use #freeswitch-gui. If this really > takes off I'm sure that we could even start a mailing list for GUI > development. > Once the goals and features decided I think more ppl can join and work this out together. > Enjoy! > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/be210a39/attachment.html From regs at kinetix.gr Fri Jun 5 01:35:52 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 05 Jun 2009 11:35:52 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> <4A2808A9.8070409@kinetix.gr> <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> Message-ID: <4A28D8E8.4040000@kinetix.gr> Anthony Minessale wrote: > FS uses async rtp timers so you may want to set rtp-timer-name=none in > the profile param to simulate asterisk conditions. I tried that - although I am not using rtp in my scenario - with the same results. > Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit > single cpu box because that was what was popular when it was designed > and the chance for race conditions is minimal because there is only 1 > cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic > difference. Yes I know that this machine is not well suited for today's test needs. But the issue occurs in every machine as long as it is pushed near (but not quite near) to its limits. I have the same odd durations using a 64 bit low end server. In this case I could achieve a better call/sec rate than that of the crappy PC but around 50-60 calls/sec the same problem showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the same thing happened at a higher rate. > > I will be happy to investigate this issue a bit if you'd like but i do > not have any box like you describe so if I can't find anything > you may have to lend us your lab. I would appreciate it if you did. After all there this might be a problem that has not surfaced yet but someday will as more and more production boxes start using FS. So it would be better to investigate it now. I don't think lending you access to my old P4 PC would help you very much :) If you have access to a normal 2-4 core system you can easily start raising the sipp parameters until it starts happening. However if you really think it is appropriate to use my test machines I'd be happy to grant access to our low-end Opteron machine (just send me a personal email). I cannot grant you access to larger systems because they are used in production. I used the embedded sipp scenarios : on the UAS side : sipp -i -mi -ci -mp 8000 -sn uas on the UAC side : sipp :5060 -s 44050505-i -mi -ci -r 70 -d 5000 -l 500 -m 2000 -sn uac The dialplan : If you need anything else from the config just notify me. In order to verify that at some point the calls start having a duration larger than the scenario's 5secs you can tcpdump on the sipp machine or turn on the cdrs logging (I know that it degrades performance, but as I said it is not a matter of when exactly it starts happening, it is a matter that it DOES start happening). > > > On Thu, Jun 4, 2009 at 12:47 PM, regs at kinetix.gr > > wrote: > > Michael Collins wrote: > > > > > > The dialplan : > > > > > > > > > > > > > > > > > expression="^.*$"> > > > > > > You forgot the parens around .* > > It should be expression="^(.*)$" if you plan to use $1 later in the > > extension... > > > > > > > > > > > data="absolute_codec_string=PCMA"/> > > > data="sofia/gateway/sipp01/$1"/> > > > > ... like here ^^^^^^^ > > :) > > -MC > > You are right! Although, I don't think that would change the outcome of > my test :) > > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From dujinfang at gmail.com Fri Jun 5 02:10:45 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 17:10:45 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> Message-ID: <5AC703CD-2434-4F1A-AFF1-0D5DAFDAF800@gmail.com> On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > I'm using openzap analog with tone_detect, it works(conference not > tested). however, according to the asterisk book, Kewlstart can > detect the busy tone and disconnect the circuit. does anyone knows > how to configure kewlstart with freeswitch/openzap? guess we don't > need tone_detect then. > > Dujinfang, > > Your telco must support "kewlstart" signaling for this to be > effective. The telco probably calls it something different, like > "disconnect supervision" or "drop in loop current" or "battery > reversal" or something like that. In any case, if the signaling is > supported then you need to set up your zaptel.conf with the > appropriate signaling type, which is either fxoks or fxsks. (I can > never remember because zaptel does it backwards where if you have an > FXO port then it uses FXS signaling but if you have an FXS port it > uses FXO signaling. Stupidity, to be sure, so be aware of it.) > > Find the sample zaptel.conf that comes with the zaptel package and > search it for fxsks or fxoks and you'll see some notes on how to set > it up for your analog trunks. > > -MC Thank you for the detailed explain MC. so the ks means kewlstart, it already set, but no luck. anyway, the tone_detect works for me, less worry about that. Thanks again. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/5cd996f5/attachment.html From wasim at convergence.pk Fri Jun 5 03:02:32 2009 From: wasim at convergence.pk (Wasim Baig) Date: Fri, 5 Jun 2009 16:02:32 +0600 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> References: <4A283DCC.5040701@gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: Just to chime in, perhaps GWT might be a good framework ... -wasim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/927e9833/attachment-0001.html From tayeb.meftah at gmail.com Fri Jun 5 03:16:13 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 05 Jun 2009 10:16:13 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: <4A28F06D.9020701@gmail.com> hello wasim, GWT will help me about Accessibility, is very very accessible tel me about how i can host GWT application thanks Wasim Baig wrote: > Just to chime in, perhaps GWT might be a good framework ... > > -wasim > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/95a19c31/attachment.html From matthew.lockwood at gmail.com Fri Jun 5 03:30:51 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Fri, 5 Jun 2009 03:30:51 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> References: <4A283DCC.5040701@gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: <415541b10906050330w35a1369al2d8fe075c38a59c0@mail.gmail.com> Thanks for all this. What I meant to respond with before I passed out asleep was: The framework doesn't matter that much. We've all got looped on this issue. Everything that is done has to add value to the end user. The framework is far down at the bottom of the list of things that provides value, but it's not something to be ignored. The vision I have for this is something that's so simple it lowers the barriers that would otherwise stop people from using FreeSwitch. Using some relatively unheard of framework is going to most certainly complicate things. Simple = good. And plus, on a side note if we throw out a whole bunch of frameworks and acronyms and make a big deal about the actual technology that powers the GUI (not that people even care most of the time), people will start to get more confused and it'll backfire. I'll be using .NET/Mono unless I can come up with an exceptionally good reason to use something else. I'm choosing this framework over everything else because it's what I know best and I'll be writing the code base. I've got years of experience writing code in C# and developing .NET web applications so it makes more sense than learning something new that will slow the development time and result in me producing poorer code. This isn't me being mercenary, but the GUI isn't likely to cross the million codeline barrier (even with everything implemented) and this is a framework I have a lot of experience with. I'm totally fine being the lone developer for now, and there are a lot of people with similar programming skillsets as mine so it's not like there will never be anybody else that'll ever contribute code. Personally, I think it's more important to have well written code that is rapidly developed than it is to have a shiny technology that adds no value. :-) Of course, the final product will be perfectly standards compliant and 100% accessible. I know this is important. I'm going to lay the framework issue to rest now. It'll be .NET/Mono unless there is some super-compelling reason to use something else. If for some reason there is such a reason not to use .NET/Mono, the second choice is PHP. The other thing is I'm pretty much going to develop upwards of 95% of the features in one go. Nobody wants an incomplete product that lacks necessary functionality, so from v1.0 it'll be pretty much feature complete. I'm developing this for use in my business, so I need it feature complete, and that's what the community will get too - a feature complete product. Hope you're happy having a fully fledged GUI! ;-) M On Fri, Jun 5, 2009 at 1:26 AM, seven wrote: > > On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: > > > > On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> That would involve me learning a totally new framework. It'll not the >> hardest code I'll ever write by far, so I'm okay coding it up on my own. >> However, I definitely need a lot of help from fabulous designers to actually >> make the interface pretty and useable. Plus, I'm only one person and will >> need a lot of feedback to create something that rocks - everybody has a >> different use case and I can't foresee how everybody will use it, so that >> kind of feedback will go into re-engineering it. >> > > If you guys are serious about this then I would like to make a few > suggestions that might be obvious but for the sake of the project we'll make > them explicitly obvious. > > First, before deciding what framework to use, it would be good to hold some > discussions about what the GUI actually needs to do: > What are the design goals? > Will it be just for setting up extensions and the dialplan? Or will it go > much farther than that? > Will you be using mod_xml_curl for everything? If so, what database(s) will > you support? > Are you going to have extra goodies like an IVR builder? > A 'visual voicemail' page? > A user portal? > Management interface to 'spy' on users? > A CDR/call accounting system? > FIFO and/or ACD queue management? > MOH and sound files management? > > > and conference management > > > > It's okay to start small and build your way out, but you need to know > before you start building what the grand scheme will be. The larger the > goals of the project, the narrower your choices for a framework that can do > it all. The simple fact of the matter is that if you want to use a MVC web > framework then you have a somewhat limited number of choices. You need a MVC > WF that fits your needs, which means it needs to be at least somewhat > flexible. If you want a pretty GUI then you need to decide if you want a > rich Internet application (RIA) front end like AIR, or do you want something > along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you > cross-browser widgets and tools. All of this on top of the fact that if you > want volunteers to assist you will need to pick something that people either > know or can learn quickly. > > Oh, and be prepared for people to give you unsolicited opinions about all > sorts of things. :) > > All that being said, I say go for it. Find what works for you and see what > happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure > that we could even start a mailing list for GUI development. > > > Once the goals and features decided I think more ppl can join and work this > out together. > > Enjoy! > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/bec4fff3/attachment.html From d at d-man.org Fri Jun 5 03:35:34 2009 From: d at d-man.org (Darren Schreiber) Date: Fri, 5 Jun 2009 03:35:34 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com><415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com><87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com><415541b10906041542t46819552s531a54db86c58754@mail.gmail.com><389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org><415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com><415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com><9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com><415541b10906041852g5441267br157501bf7297b313@mail.gmail.com><87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: This is a really ironic post, Seven. :-) I agree with all your points. A while ago I started the TCAPI project to build a front-end for FreeSWITCH. I very quickly got inundated with debates about framework and language. These debates were initially appreciated but at some point we needed to decide & move on. The real work to be done was, as you point out, in design of the application business logic, interface and actually coding it up and putting it together. So we decided to go a bit radio silent and and focus on a few developers who were willing to build out the foundational pieces of the MVC architecture, and to let you create FreeSWITCH config files and general database and software modules with a set of standardized, simple to use libraries/APIs. Once we are done with that, the intention was to release it to those who wanted to help build the pieces related to modules in FreeSWITCH. That project is about 6 weeks from release into beta, give or take a few weeks (hey, it's software dev! heh who's ever on time?). So anyone who is on here reading this and might be interested in contributing code to an already very active FreeSWITCH GUI development project please feel free to contact me - we are now accepting serious developer inquiries. The project is in PHP and uses two pretty nifty frameworks (we, as you point out, couldn't find exactly what we were looking for, so we merged two libraries that fit the bill very nicely). It is database agnostic and is designed to work on Windows or Linux so don't let that be a barrier to participation. This will be an open source project for all, btw. I will be presenting on it at the upcoming ClueCon, warts and all, so you should go register and then you can participate in the demo/tutorial! :-) - Darren _____ From: seven [mailto:dujinfang at gmail.com] Sent: Friday, June 05, 2009 1:26 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] WikiPBX Installation On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood wrote: That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? and conference management It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited opinions about all sorts of things. :) All that being said, I say go for it. Find what works for you and see what happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure that we could even start a mailing list for GUI development. Once the goals and features decided I think more ppl can join and work this out together. Enjoy! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/9230ded1/attachment-0001.html From larclap at yahoo.com Fri Jun 5 06:00:44 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 5 Jun 2009 06:00:44 -0700 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: References: <079f01c9e535$c1fed2e0$45fc78a0$@com> <08bc01c9e57e$5ef60030$1ce20090$@com> Message-ID: <09a501c9e5dd$a36e2e90$ea4a8bb0$@com> Thanks, Jim, that advice really helped in more ways than I asked for. Lars -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim Burke Sent: Thursday, June 04, 2009 6:52 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing lines copying data from console to vi After logging into my linux box using putty. I then change directory the the ~/freeswitch/bin directory and run ./fs_cli You can, but you don't need to start FS from the putty terminal. We run ours as a background process. On Fri, Jun 5, 2009 at 11:38 AM, Lars Zeb wrote: > Does this mean that I must start fs from the putty terminal, or can I attach > to an already running instance via putty? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim > Burke > Sent: Thursday, June 04, 2009 4:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Missing lines copying data from console to > vi > > Not that this helps your question directly. ?Using a putty terminal > from windows allows the data to be copied correctly from console > sessions. > > > > On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: >> I want to copy the results of a siptrace captured on the fs console to a >> file. The console is running on a Gnome terminal. I highlight the text I >> want to copy in the fs console, open a vi session in insert mode, and > paste >> the text. However, the text is not pasted as I copied it - it is missing >> characters/lines. >> >> >> >> I know I am doing something wrong. Is there another way to save siptraces > to >> a file? Redirection doesn't work. >> >> >> >> sofia profile internal siptrace on is the command I use. >> >> >> >> Thanks Lars >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From shoaib at breezecom.ae Thu Jun 4 22:35:13 2009 From: shoaib at breezecom.ae (Shoaib Khanzada) Date: Fri, 5 Jun 2009 11:35:13 +0600 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: <001b01c9e59f$68ffdfd0$3aff9f70$@ae> Hi FS-Users, I am having an strange problem of sessions in freeswitch. For every call Freeswitch is creating more than two sessions (in and out legs). For example, I have seen 260+ sessions for just 30+ calls whereas there should not be more than 60 sessions for just 30 calls. I have seen the same problem with 1.0.4pre8 and trunk version. I am using default values from switch.conf.xml for max_sessions (1000) and sessions_per_second(30). Freeswitch create many sessions only when there is good load on the system. It works fine with the steady load of 50-100 calls. However, if I give it 200+ calls at once then it breaks. Any suggestion? Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/ecc46a4a/attachment.html From brian at freeswitch.org Fri Jun 5 07:07:13 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 09:07:13 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: <3wmhHdntwTK5.tvBsy3CM@smtp.gmail.com> References: <3wmhHdntwTK5.tvBsy3CM@smtp.gmail.com> Message-ID: DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > should open a jira on this? Did you guys need any more info? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/50d872ab/attachment.html From anthony.minessale at gmail.com Fri Jun 5 07:08:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Jun 2009 09:08:09 -0500 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: <001b01c9e59f$68ffdfd0$3aff9f70$@ae> References: <001b01c9e59f$68ffdfd0$3aff9f70$@ae> Message-ID: <191c3a030906050708q7fc62c74j896b08f0454864d1@mail.gmail.com> No real suggestions because there is very little information here. We tend to not address load testing issues because most problems are caused by improper test environments and user error. My suggestion is to try it for real instead of in a test land. On Fri, Jun 5, 2009 at 12:35 AM, Shoaib Khanzada wrote: > Hi FS-Users, > > > > I am having an strange problem of sessions in freeswitch. > > > > For every call Freeswitch is creating more than two sessions (in and out > legs). For example, I have seen 260+ sessions for just 30+ calls whereas > there should not be more than 60 sessions for just 30 calls. > > > > I have seen the same problem with 1.0.4pre8 and trunk version. > > > > I am using default values from switch.conf.xml for max_sessions (1000) and > sessions_per_second(30). > > > > Freeswitch create many sessions only when there is good load on the system. > It works fine with the steady load of 50-100 calls. However, if I give it > 200+ calls at once then it breaks. > > > > Any suggestion? > > > > Shoaib > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/e1d3db74/attachment.html From msc at freeswitch.org Fri Jun 5 09:27:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 5 Jun 2009 09:27:37 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: <87f2f3b90906050927t6022fab9ka1718e8cec98727c@mail.gmail.com> On Fri, Jun 5, 2009 at 3:35 AM, Darren Schreiber wrote: > This is a really ironic post, Seven. :-) I agree with all your points. > > A while ago I started the TCAPI project to build a front-end for > FreeSWITCH. I very quickly got inundated with debates about framework and > language. These debates were initially appreciated but at some point we > needed to decide & move on. The real work to be done was, as you point out, > in design of the application business logic, interface and actually coding > it up and putting it together. So we decided to go a bit radio silent and > and focus on a few developers who were willing to build out the foundational > pieces of the MVC architecture, and to let you create FreeSWITCH config > files and general database and software modules with a set of standardized, > simple to use libraries/APIs. Once we are done with that, the intention was > to release it to those who wanted to help build the pieces related to > modules in FreeSWITCH. That project is about 6 weeks from release into > beta, give or take a few weeks (hey, it's software dev! heh who's ever on > time?). > > I was wondering when you were gonna chime in on this subject! :D -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/6c223767/attachment-0001.html From marcus.frenkel at gmail.com Fri Jun 5 11:14:15 2009 From: marcus.frenkel at gmail.com (Marcus Frenkel) Date: Fri, 5 Jun 2009 20:14:15 +0200 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH Message-ID: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> Hi, I'm using GnuGK H323 gatekeeper. It has good performance and many features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch comparisons, but how about GnuGK vs FreeSwitch? The setup for which I'm asking is in a range of 200 concurrent calls. The points that I'm really interested for comparison are: 1) Proxy of RTP feature and it's stability 2) NAT support 3) Direct SQL AAA support (without the need of using RADIUS server) 4) Performance as an endpoint registrar 5) Rerouting to a second carrier on failed call Also, is the H323 library of FreeSWITCH based on h323plus/openh323? Marcus From anthony.minessale at gmail.com Fri Jun 5 12:47:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Jun 2009 14:47:45 -0500 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH In-Reply-To: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> References: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> Message-ID: <191c3a030906051247m542efe08l2011dd25c0e81c34@mail.gmail.com> the h323 is done with mod_opal On Fri, Jun 5, 2009 at 1:14 PM, Marcus Frenkel wrote: > Hi, > > I'm using GnuGK H323 gatekeeper. It has good performance and many > features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch > comparisons, but how about GnuGK vs FreeSwitch? > > The setup for which I'm asking is in a range of 200 concurrent calls. > > The points that I'm really interested for comparison are: > 1) Proxy of RTP feature and it's stability > 2) NAT support > 3) Direct SQL AAA support (without the need of using RADIUS server) > 4) Performance as an endpoint registrar > 5) Rerouting to a second carrier on failed call > > Also, is the H323 library of FreeSWITCH based on h323plus/openh323? > > Marcus > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/6f29d867/attachment.html From edpimentl at gmail.com Fri Jun 5 13:11:28 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 5 Jun 2009 16:11:28 -0400 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH In-Reply-To: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> References: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> Message-ID: <9dc4a1670906051311m7bf6a7acw9ae9d8f327ef6d19@mail.gmail.com> Is this a joke or do want to save the question for April Fools? -E On Fri, Jun 5, 2009 at 2:14 PM, Marcus Frenkel wrote: > Hi, > > I'm using GnuGK H323 gatekeeper. It has good performance and many > features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch > comparisons, but how about GnuGK vs FreeSwitch? > > The setup for which I'm asking is in a range of 200 concurrent calls. > > The points that I'm really interested for comparison are: > 1) Proxy of RTP feature and it's stability > 2) NAT support > 3) Direct SQL AAA support (without the need of using RADIUS server) > 4) Performance as an endpoint registrar > 5) Rerouting to a second carrier on failed call > > Also, is the H323 library of FreeSWITCH based on h323plus/openh323? > > Marcus > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/94dd9710/attachment.html From skhanzada at gmail.com Fri Jun 5 13:09:54 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 02:09:54 +0600 Subject: [Freeswitch-users] Reuse ODBC connection in javascript Message-ID: <8eace5520906051309l4916a1d9p3c95497df0f0b17f@mail.gmail.com> Hi, I want to reuse database connection (mysql) in one of my javascript which is executed on each call. Following is how I am creating ODBC connection. Line1) var db = new ODBC(DSN, DB_USER, DB_PASS); Line2) db.connect(); My first question is, where does it create a database connection on line1 or on line2? Secondly, how can I reuse this connection? so that it is not created for each call and I just use a previously created db object in my routing script. My Objective is: 1) Create ODBC connection on freeswitch startup (could be a database connection pool) 2) Reuse the connection on each call 3) Close database connection on freeswitch shutdown (or on some other event) Any help would be highly appreciated. Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/431b0b94/attachment-0001.html From skhanzada at gmail.com Fri Jun 5 13:10:21 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 02:10:21 +0600 Subject: [Freeswitch-users] Using setGlobalVar and getGlobalVar Message-ID: <8eace5520906051310i79f3d037lcb1765ef2ac158b4@mail.gmail.com> Hi, How can I use setGlobalVar and getGlobalVar in my javascript to store a ODBC connection? I want to set an ODBC database connection object globally so that I can access it from anywhere. This connection will be used for read-only so no threading issues. Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/764e7778/attachment-0001.html From skhanzada at gmail.com Fri Jun 5 13:11:15 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 02:11:15 +0600 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: <8eace5520906051311u19d36449scd447b05975b605d@mail.gmail.com> Thanks for the reply?. I was not running it for testing purpose because I?ve completed all testing successfully. We are running freeswitch in a voip carrier grade environment. It works perfectly alright in off peak hours when we have low calls ratio around 100-150 concurrent calls. However, in peak hours this goes beyond 200 calls and on that point freeswitch start creating many sessions for each calls. I have seen the sessions using ?status? command and calls using ?show calls count?. We are using javascript to select the route from the mysql database for each call. Could it be because script is taking longer than expected amount of time to retrieve a route? and freeswitch continuously keep creating sessions for incoming calls. That?s why I see low no of connected calls (if ?show calls count? only display the connected calls) whereas sessions are continuously being created by freeswitch as it is receiving many calls. If above text confuses you, nevermind just answer the following questions. 1) Does ?show calls count? display the connected calls only? 2) When freeswitch create session instances? Before bridge or after bridge? Or one before bridge and one after bridge? Thanks, Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/48b934e6/attachment-0001.html From mike at jerris.com Fri Jun 5 13:23:23 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Jun 2009 16:23:23 -0400 Subject: [Freeswitch-users] Using setGlobalVar and getGlobalVar In-Reply-To: <8eace5520906051310i79f3d037lcb1765ef2ac158b4@mail.gmail.com> References: <8eace5520906051310i79f3d037lcb1765ef2ac158b4@mail.gmail.com> Message-ID: On Jun 5, 2009, at 4:10 PM, Shoaib Khanzada wrote: > Hi, > > How can I use setGlobalVar and getGlobalVar in my javascript to > store a ODBC connection? > > I want to set an ODBC database connection object globally so that I > can access it from anywhere. This connection will be used for read- > only so no threading issues. > no, those are for strings only. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/e0142ff9/attachment.html From mike at jerris.com Fri Jun 5 13:25:38 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Jun 2009 16:25:38 -0400 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: <8eace5520906051311u19d36449scd447b05975b605d@mail.gmail.com> References: <8eace5520906051311u19d36449scd447b05975b605d@mail.gmail.com> Message-ID: <40DD2D82-68BF-4898-8233-636D6EBB2E22@jerris.com> On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: > Thanks for the reply?. > > I was not running it for testing purpose because I?ve completed all > testing successfully. > > We are running freeswitch in a voip carrier grade environment. It > works perfectly alright in off peak hours when we have low calls > ratio around 100-150 concurrent calls. However, in peak hours this > goes beyond 200 calls and on that point freeswitch start creating > many sessions for each calls. I have seen the sessions using > ?status? command and calls using ?show calls count?. > > We are using javascript to select the route from the mysql database > for each call. Could it be because script is taking longer than > expected amount of time to retrieve a route? and freeswitch > continuously keep creating sessions for incoming calls. That?s why I > see low no of connected calls (if ?show calls count? only display > the connected calls) whereas sessions are continuously being created > by freeswitch as it is receiving many calls. > > If above text confuses you, nevermind just answer the following > questions. > > 1) Does ?show calls count? display the connected calls only? Only bridged calls (2 sessions) > 2) When freeswitch create session instances? Before bridge or > after bridge? Or one before bridge and one after bridge? It creates a session when it gets an incomming call and creates one for each outgoing call, unrelated to bridging. > > Thanks, > > Shoaib How are you doing the bridge in your script? Are you setting a var then dropping out of the js to do the bridge? Can you post your js file? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/0b060607/attachment.html From marcus.frenkel at gmail.com Fri Jun 5 14:55:14 2009 From: marcus.frenkel at gmail.com (Marcus Frenkel) Date: Fri, 5 Jun 2009 23:55:14 +0200 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH In-Reply-To: <9dc4a1670906051311m7bf6a7acw9ae9d8f327ef6d19@mail.gmail.com> References: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> <9dc4a1670906051311m7bf6a7acw9ae9d8f327ef6d19@mail.gmail.com> Message-ID: <1ded54630906051455r60ed2194he8b1ad22f64d4f63@mail.gmail.com> Can you explain your friendly advice? Some parts can be compared On Fri, Jun 5, 2009 at 10:11 PM, EdPimentl wrote: > Is this a joke or do want to save the question for April Fools? > -E > > > On Fri, Jun 5, 2009 at 2:14 PM, Marcus Frenkel > wrote: >> >> Hi, >> >> I'm using GnuGK H323 gatekeeper. It has good performance and many >> features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch >> comparisons, but how about GnuGK vs FreeSwitch? >> >> The setup for which I'm asking is in a range of 200 concurrent calls. >> >> The points that I'm really interested for comparison are: >> 1) Proxy of RTP feature and it's stability >> 2) NAT support >> 3) Direct SQL AAA support (without the need of using RADIUS server) >> 4) Performance as an endpoint registrar >> 5) Rerouting to a second carrier on failed call >> >> Also, is the H323 library of FreeSWITCH based on h323plus/openh323? >> >> Marcus >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jim at evolutiontel.net Fri Jun 5 16:25:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Sat, 6 Jun 2009 09:25:39 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. Message-ID: Yup sure did, same result :( - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Brian West Date: 05/06/2009 14:10 DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > should open a jira on this? Did you guys need any more info? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Jun 5 16:53:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Jun 2009 18:53:02 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: References: Message-ID: <191c3a030906051653nd8c56d4wb50a6ab64e1a259f@mail.gmail.com> it is using G729a which is not the correct RFC value that goes with payload 18 so it's moving it to 96 as if it's some other codec. the only thing you can do is get them to stop using invalid data in their sdp or hack it to replace "G729a" with "G729 " before it's too late. On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/777cc863/attachment.html From mrene_lists at avgs.ca Fri Jun 5 16:54:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 5 Jun 2009 19:54:48 -0400 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: <191c3a030906051653nd8c56d4wb50a6ab64e1a259f@mail.gmail.com> References: <191c3a030906051653nd8c56d4wb50a6ab64e1a259f@mail.gmail.com> Message-ID: <68EDAD18-27B1-4BB5-86E1-573A55F0D72C@avgs.ca> Linksys still uses G729a in their sdp, but you can change it in the admin panel (if thats what you have) Math On 5-Jun-09, at 7:53 PM, Anthony Minessale wrote: > it is using G729a which is not the correct RFC value that goes with > payload 18 so it's moving it to 96 as if it's some other > codec. > > the only thing you can do is get them to stop using invalid data in > their sdp > or hack it to replace "G729a" with "G729 " before it's too late. > > > On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke > wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/bf1ba320/attachment-0001.html From jim at evolutiontel.net Fri Jun 5 17:00:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Sat, 6 Jun 2009 10:00:39 +1000 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: Could it be that you are getting several INVITE messages per call due to 100 Trying message is not being sent until after the route is selected from your java routing script? You might be able to send a 100 trying from your dialplan or script. Keep in mind that this message stops timers on the originating side. - original message - Subject: Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call From: Michael Jerris Date: 05/06/2009 20:27 On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: > Thanks for the reply?. > > I was not running it for testing purpose because I?ve completed all > testing successfully. > > We are running freeswitch in a voip carrier grade environment. It > works perfectly alright in off peak hours when we have low calls > ratio around 100-150 concurrent calls. However, in peak hours this > goes beyond 200 calls and on that point freeswitch start creating > many sessions for each calls. I have seen the sessions using > ?status? command and calls using ?show calls count?. > > We are using javascript to select the route from the mysql database > for each call. Could it be because script is taking longer than > expected amount of time to retrieve a route? and freeswitch > continuously keep creating sessions for incoming calls. That?s why I > see low no of connected calls (if ?show calls count? only display > the connected calls) whereas sessions are continuously being created > by freeswitch as it is receiving many calls. > > If above text confuses you, nevermind just answer the following > questions. > > 1) Does ?show calls count? display the connected calls only? Only bridged calls (2 sessions) > 2) When freeswitch create session instances? Before bridge or > after bridge? Or one before bridge and one after bridge? It creates a session when it gets an incomming call and creates one for each outgoing call, unrelated to bridging. > > Thanks, > > Shoaib How are you doing the bridge in your script? Are you setting a var then dropping out of the js to do the bridge? Can you post your js file? Mike _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Fri Jun 5 19:11:58 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 6 Jun 2009 10:11:58 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> Message-ID: <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > I'm using openzap analog with tone_detect, it works(conference not > tested). however, according to the asterisk book, Kewlstart can > detect the busy tone and disconnect the circuit. does anyone knows > how to configure kewlstart with freeswitch/openzap? guess we don't > need tone_detect then. > > Dujinfang, > > Your telco must support "kewlstart" signaling for this to be > effective. The telco probably calls it something different, like > "disconnect supervision" or "drop in loop current" or "battery > reversal" or something like that. In any case, if the signaling is > supported then you need to set up your zaptel.conf with the > appropriate signaling type, which is either fxoks or fxsks. (I can > never remember because zaptel does it backwards where if you have an > FXO port then it uses FXS signaling but if you have an FXS port it > uses FXO signaling. Stupidity, to be sure, so be aware of it.) > 1) Don't know why but the similar zaptel.conf works on asterisk. I guess tone_detect in FS is equivalent to busydetect=yes in Asterisk(zapata.conf) . zaptel.conf # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" fxoks=5 fxoks=6 fxoks=7 fxoks=8 # Global data loadzone = us defaultzone = us zapata.conf usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes I agree the FXO and FXS signaling is weird, why not they just match the care name and reverse that internally? 2) Another essue is if I dial out from a FXO port from a local extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo on asterisk. the zt.conf as below and I tried to change the echo_cancel_level to 32 or 128 got no much difference. Is there any equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? can I set busydetect and echocancelwhenbridged and other options like this ? [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 > Find the sample zaptel.conf that comes with the zaptel package and > search it for fxsks or fxoks and you'll see some notes on how to set > it up for your analog trunks. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/d6b9464d/attachment.html From gcd at i.ph Fri Jun 5 19:24:09 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 6 Jun 2009 10:24:09 +0800 Subject: [Freeswitch-users] Reducing record_session load Message-ID: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> we experience some latency in the recording files even with PCMU-PCMU session to a stereo WAV file. i want to reduce the CPU load hoping to reduce this problem. would it help if do the ff? 1. save it in PCMU file. i can use sox at the end of the shift. 2. record in mono. does it help? 3. will record_session work w/ proxy_media=true? tks for your help. -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/88e9f6f4/attachment.html From jim at evolutiontel.net Fri Jun 5 19:25:32 2009 From: jim at evolutiontel.net (Jim Burke) Date: Sat, 6 Jun 2009 12:25:32 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. Message-ID: Yes, the terminator is Linksys so will change it and test. Noticed there is a list of mime types associated with FS and G729a was not listed, does this have anything to do with the root cause? - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Mathieu Rene Date: 05/06/2009 23:56 Linksys still uses G729a in their sdp, but you can change it in the admin panel (if thats what you have) Math On 5-Jun-09, at 7:53 PM, Anthony Minessale wrote: > it is using G729a which is not the correct RFC value that goes with > payload 18 so it's moving it to 96 as if it's some other > codec. > > the only thing you can do is get them to stop using invalid data in > their sdp > or hack it to replace "G729a" with "G729 " before it's too late. > > > On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke > wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Fri Jun 5 19:32:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 21:32:01 -0500 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> Message-ID: On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote: > we experience some latency in the recording files even with PCMU- > PCMU session to a stereo WAV file. i want to reduce the CPU load > hoping to reduce this problem. would it help if do the ff? > 1. save it in PCMU file. i can use sox at the end of the shift. You shouldn't be experiencing this at all... how many are you doing at once? > 2. record in mono. does it help? No. > 3. will record_session work w/ proxy_media=true? Nope. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/fa3c9cdb/attachment.html From brian at freeswitch.org Fri Jun 5 19:32:37 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 21:32:37 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: References: Message-ID: <51DECC6F-E822-4459-BF47-970576041825@freeswitch.org> G729a is 100% INVALID in the sdp on codec 18. /b On Jun 5, 2009, at 9:25 PM, Jim Burke wrote: > Noticed there is a list of mime types associated with FS and G729a > was not listed, does this have anything to do with the root cause? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/95ea73ad/attachment-0001.html From gcd at i.ph Fri Jun 5 19:58:05 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 6 Jun 2009 10:58:05 +0800 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> Message-ID: <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> there 10 client seats so at max. 10 simultaneous calls. however, the number of clients may be increased. -nandy On Sat, Jun 6, 2009 at 10:32 AM, Brian West wrote: > > On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote: > > we experience some latency in the recording files even with PCMU-PCMU > session to a stereo WAV file. i want to reduce the CPU load hoping to reduce > this problem. would it help if do the ff? > 1. save it in PCMU file. i can use sox at the end of the shift. > > > You shouldn't be experiencing this at all... how many are you doing at > once? > > 2. record in mono. does it help? > > > No. > > 3. will record_session work w/ proxy_media=true? > > > Nope. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/e0e8af65/attachment.html From brian at freeswitch.org Fri Jun 5 20:02:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 22:02:53 -0500 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> Message-ID: <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> You shouldn't be having problems... what version are you using? /b On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: > there 10 client seats so at max. 10 simultaneous calls. however, the > number of clients may be increased. > -nandy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/39558777/attachment.html From gcd at i.ph Fri Jun 5 20:03:20 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 6 Jun 2009 11:03:20 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> Message-ID: <7d0bfd8c0906052003w40a8fff4s8644cb9773e346b5@mail.gmail.com> dujinfang, hv u tried OSLEC? it's really reduced echo even on the cheapy X100P card on *. oslec works w/ FS, too. -nandy On Sat, Jun 6, 2009 at 10:11 AM, dujinfang wrote: > > On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: > > > > On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > >> I'm using openzap analog with tone_detect, it works(conference not >> tested). however, according to the asterisk book, Kewlstart can detect the >> busy tone and disconnect the circuit. does anyone knows how to configure >> kewlstart with freeswitch/openzap? guess we don't need tone_detect then. >> > > Dujinfang, > > Your telco must support "kewlstart" signaling for this to be effective. The > telco probably calls it something different, like "disconnect supervision" > or "drop in loop current" or "battery reversal" or something like that. In > any case, if the signaling is supported then you need to set up your > zaptel.conf with the appropriate signaling type, which is either fxoks or > fxsks. (I can never remember because zaptel does it backwards where if you > have an FXO port then it uses FXS signaling but if you have an FXS port it > uses FXO signaling. Stupidity, to be sure, so be aware of it.) > > > 1) Don't know why but the similar zaptel.conf works on asterisk. I guess > tone_detect in FS is equivalent to busydetect=yes in > Asterisk(zapata.conf) . > > > zaptel.conf > # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) > fxsks=1 > fxsks=2 > fxsks=3 > fxsks=4 > > # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" > fxoks=5 > fxoks=6 > fxoks=7 > fxoks=8 > > # Global data > > loadzone = us > defaultzone = us > > > zapata.conf > > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=no > ;echotraining=800 > rxgain=0.0 > txgain=0.0 > group=0 > callgroup=1 > pickupgroup=1 > immediate=no > busydetect=yes > > I agree the FXO and FXS signaling is weird, why not they just match the > care name and reverse that internally? > > 2) Another essue is if I dial out from a FXO port from a local > extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo > on asterisk. the zt.conf as below and I tried to change the > echo_cancel_level to 32 or 128 got no much difference. Is there any > equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? > can I set busydetect and echocancelwhenbridged and other options like > this ? > > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > echo_cancel_level => 64 > > > Find the sample zaptel.conf that comes with the zaptel package and search > it for fxsks or fxoks and you'll see some notes on how to set it up for your > analog trunks. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/de5569e5/attachment.html From mike at jerris.com Fri Jun 5 20:17:03 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Jun 2009 23:17:03 -0400 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: References: Message-ID: We already send a 100 before the call even hits the dialplan. Mike On Jun 5, 2009, at 8:00 PM, "Jim Burke" wrote: > Could it be that you are getting several INVITE messages per call > due to 100 Trying message is not being sent until after the route is > selected from your java routing script? > > You might be able to send a 100 trying from your dialplan or script. > Keep in mind that this message stops timers on the originating side. > > - original message - > Subject: Re: [Freeswitch-users] Freeswitch creating more then two > sessions for one call > From: Michael Jerris > Date: 05/06/2009 20:27 > > > On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: > >> Thanks for the reply?. >> >> I was not running it for testing purpose because I?ve completed all >> testing successfully. >> >> We are running freeswitch in a voip carrier grade environment. It >> works perfectly alright in off peak hours when we have low calls >> ratio around 100-150 concurrent calls. However, in peak hours this >> goes beyond 200 calls and on that point freeswitch start creating >> many sessions for each calls. I have seen the sessions using >> ?status? command and calls using ?show calls count?. >> >> We are using javascript to select the route from the mysql database >> for each call. Could it be because script is taking longer than >> expected amount of time to retrieve a route? and freeswitch >> continuously keep creating sessions for incoming calls. That?s why I >> see low no of connected calls (if ?show calls count? only display >> the connected calls) whereas sessions are continuously being created >> by freeswitch as it is receiving many calls. >> >> If above text confuses you, nevermind just answer the following >> questions. >> >> 1) Does ?show calls count? display the connected calls only? > > Only bridged calls (2 sessions) > >> 2) When freeswitch create session instances? Before bridge or >> after bridge? Or one before bridge and one after bridge? > > It creates a session when it gets an incomming call and creates one > for each outgoing call, unrelated to bridging. > >> >> Thanks, >> >> Shoaib > > How are you doing the bridge in your script? Are you setting a var > then dropping out of the js to do the bridge? Can you post your js > file? > > Mike > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From klaus.teller at gmx.net Fri Jun 5 20:34:26 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sat, 06 Jun 2009 05:34:26 +0200 Subject: [Freeswitch-users] How to reject a call without answering Message-ID: <20090606033426.72340@gmx.net> Hi, Going through the socket api, how can i reject a call without having to answer it first? I tried sending a hangup command with cause set either to NO_ANSWER or NORMAL_CLEARING. In both cases, Freeswitch does create another socket to deliver the very same call. More precisely, when a call comes in i send a connect command. Then after some few seconds, i then send the following hangup command: SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f call-command: hangup hangup-cause: NO_ANSWER Thanks for any feedback. Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From brian at freeswitch.org Fri Jun 5 20:45:29 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 22:45:29 -0500 Subject: [Freeswitch-users] How to reject a call without answering In-Reply-To: <20090606033426.72340@gmx.net> References: <20090606033426.72340@gmx.net> Message-ID: <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> Try the respond app. /b On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: > Hi, > > Going through the socket api, how can i reject a call without having > to answer it first? > > I tried sending a hangup command with cause set either to NO_ANSWER > or NORMAL_CLEARING. In both cases, Freeswitch does create another > socket to deliver the very same call. > > More precisely, when a call comes in i send a connect command. Then > after some few seconds, i then send the following hangup command: > > SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f > call-command: hangup > hangup-cause: NO_ANSWER > > Thanks for any feedback. > > Klaus. > > -- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/055745ec/attachment.html From klaus.teller at gmx.net Fri Jun 5 21:28:46 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sat, 06 Jun 2009 06:28:46 +0200 Subject: [Freeswitch-users] How to reject a call without answering In-Reply-To: <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> References: <20090606033426.72340@gmx.net> <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> Message-ID: <20090606042846.72320@gmx.net> It doesn't seem to work. I tried the following: api respond 9015430e-82cf-418c-bf4c-f3ac6e85caf2 503 SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 call-command: execute execute-app-name: respond execute-app-arg: 503 Is one of these what you meant? Klaus. -------- Original-Nachricht -------- > Datum: Fri, 5 Jun 2009 22:45:29 -0500 > Von: Brian West > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] How to reject a call without answering > Try the respond app. > > /b > > On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: > > > Hi, > > > > Going through the socket api, how can i reject a call without having > > to answer it first? > > > > I tried sending a hangup command with cause set either to NO_ANSWER > > or NORMAL_CLEARING. In both cases, Freeswitch does create another > > socket to deliver the very same call. > > > > More precisely, when a call comes in i send a connect command. Then > > after some few seconds, i then send the following hangup command: > > > > SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f > > call-command: hangup > > hangup-cause: NO_ANSWER > > > > Thanks for any feedback. > > > > Klaus. > > > > -- > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > -- GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 Euro/mtl.! http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a From jmesquita at gmail.com Fri Jun 5 21:47:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 6 Jun 2009 01:47:50 -0300 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906050927t6022fab9ka1718e8cec98727c@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> <87f2f3b90906050927t6022fab9ka1718e8cec98727c@mail.gmail.com> Message-ID: <5a8712120906052147v45962d5r6539d40b153e89e7@mail.gmail.com> Hail to someone who has actually done something! :-) Darren, I am excited to started working on that with you all. Maybe adding mod_khomp support to it whenever I have some actual working code. jmesquita On Fri, Jun 5, 2009 at 1:27 PM, Michael Collins wrote: > > > On Fri, Jun 5, 2009 at 3:35 AM, Darren Schreiber wrote: > >> This is a really ironic post, Seven. :-) I agree with all your points. >> >> A while ago I started the TCAPI project to build a front-end for >> FreeSWITCH. I very quickly got inundated with debates about framework and >> language. These debates were initially appreciated but at some point we >> needed to decide & move on. The real work to be done was, as you point out, >> in design of the application business logic, interface and actually coding >> it up and putting it together. So we decided to go a bit radio silent and >> and focus on a few developers who were willing to build out the foundational >> pieces of the MVC architecture, and to let you create FreeSWITCH config >> files and general database and software modules with a set of standardized, >> simple to use libraries/APIs. Once we are done with that, the intention was >> to release it to those who wanted to help build the pieces related to >> modules in FreeSWITCH. That project is about 6 weeks from release into >> beta, give or take a few weeks (hey, it's software dev! heh who's ever on >> time?). >> >> > > I was wondering when you were gonna chime in on this subject! :D > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/e792228f/attachment.html From mitul at enterux.com Fri Jun 5 22:23:30 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 6 Jun 2009 10:53:30 +0530 Subject: [Freeswitch-users] (no subject) Message-ID: <182EFB0F-8623-4A1B-A457-D171EC805E9F@enterux.com> Ttrfrtttgteruoywtklou Regards,juuyuuu Mitul Limbani, Founder & CEO, iuiiiiokljkkllllllllllllmmmmnnvvfcxzasqwwhjhyljljjifkkkljjyjjjkkjllgjjggllyjkljkokjkjjjujjjjjkmktdswwdsflyjhhhhhhbhh mmmmmlkkkjjjhhhjykvytyyp Enterux Solutions Pvt Ltd,bu. B. P The Enterprise Linux Company(r), http://www.enterux.com/i Pio From gmaruzz at celliax.org Fri Jun 5 23:23:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 6 Jun 2009 08:23:26 +0200 Subject: [Freeswitch-users] (no subject) In-Reply-To: <182EFB0F-8623-4A1B-A457-D171EC805E9F@enterux.com> References: <182EFB0F-8623-4A1B-A457-D171EC805E9F@enterux.com> Message-ID: <7b197bef0906052323m56ba507y75829094293f4771@mail.gmail.com> I agree! Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sat, Jun 6, 2009 at 7:23 AM, Mitul Limbani wrote: > Ttrfrtttgteruoywtklou > > Regards,juuyuuu > Mitul Limbani, > Founder & > CEO, > iuiiiiokljkkllllllllllllmmmmnnvvfcxzasqwwhjhyljljjifkkkljjyjjjkkjllgjjggllyjkljkokjkjjjujjjjjkmktdswwdsflyjhhhhhhbhh > mmmmmlkkkjjjhhhjykvytyyp > Enterux Solutions Pvt Ltd,bu. B. ?P > The Enterprise Linux Company(r), > http://www.enterux.com/i > Pio > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From skhanzada at gmail.com Sat Jun 6 00:53:59 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 13:53:59 +0600 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: <8eace5520906060053s70ecd641y269834ec23d045b7@mail.gmail.com> Following is the js i am using to select route and bridge the call.... use("ODBC"); var DSN = "myodbc"; var DB_USER = "neo"; var DB_PASS = "....."; var sql; var prefix1; var ip1; var no2; var bridge_str; if(argv[0]=="") exit(); if(argv[1]=="") exit(); ip1 =argv[0].split("@")[1]; var no1=argv[1]; var sql = "SELECT id,prefix,name FROM internal_auth where substring('"+no1+"',1,length(prefix))=prefix and ip='" + ip1 +"' and active=1;"; var authName=""; var db = new ODBC(DSN, DB_USER, DB_PASS); db.connect(); db.query(sql); if(db.nextRow()) { var row = db.getData(); if (row["id"] == "" ) { console_log("Err", "auth failed(0001)\t" + no1 + "\targ[0]=>" + argv[0] + "\targ[1]=>" + argv[1] + "\n"); db.close(); exit(); } prefix1=row["prefix"]; authName=row["name"]; }else{ console_log("Err", "auth failed(0001)\t" + no1 + "\targ[0]=>" + argv[0] + "\targ[1]=>" + argv[1] + "\n"); db.close(); exit(); } no2=no1.substring(prefix1.length, no1.length); sql = "SELECT ec.* FROM auth_routes_carriers arc,external_carriers ec, codes c where arc.external_carriers_id = ec.id and arc.code_id = c.id and substring('"+no2+"',1,length(c.code))=c.code and ec.active=1 order by arc.priority asc"; db.connect(); db.query(sql); var carriersName=""; bridge_str=''; while(db.nextRow()) { var row = db.getData(); if (row["id"] != "" ) { bridge_str = row["gateway"]+row["prefix"]+no2+"\@"+row["ip"]; carriersName = row["name"]; break; } } db.close(); if(bridge_str != ''){ if(session.ready()) session.execute("bridge", bridge_str); } else { console_log("Err", "external carrier not found.\t" + no2 + "\targ[0]=>" + argv[0] + "\targ[1]=>" + argv[1] + "\n"); exit(); } exit(); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/f73b2bae/attachment.html From rupa at rupa.com Sat Jun 6 06:26:05 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 6 Jun 2009 08:26:05 -0500 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: <8eace5520906060053s70ecd641y269834ec23d045b7@mail.gmail.com> References: <8eace5520906060053s70ecd641y269834ec23d045b7@mail.gmail.com> Message-ID: Some thoughts: 1) look at mod_lcr 2) comments on script below... On Sat, Jun 6, 2009 at 2:53 AM, Shoaib Khanzada wrote: > Following is the js i am using to select route and bridge the call.... > > > use("ODBC"); > > var DSN = "myodbc"; > var DB_USER = "neo"; > var DB_PASS = "....."; > > [...] > var db = new ODBC(DSN, DB_USER, DB_PASS); > > db.connect(); > > [...] > > sql = "SELECT ec.* FROM auth_routes_carriers arc,external_carriers ec, > codes c where arc.external_carriers_id = ec.id and arc.code_id = c.id and > substring('"+no2+"',1,length(c.code))=c.code and ec.active=1 order by > arc.priority asc"; > > db.connect(); > no need to reconnect > > db.query(sql); > > [...] > > if(bridge_str != ''){ > if(session.ready()) session.execute("bridge", bridge_str); > Don't actually execute the bridge from javascript. Instead, set the bridge_str to a channel var and then do the bridge from the dialplan using that bridge_str. This way you don't have a javascript interpreter lying around for the duration of the call. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/868ef27d/attachment-0001.html From nik.middleton at noblesolutions.co.uk Sat Jun 6 10:39:13 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 6 Jun 2009 18:39:13 +0100 Subject: [Freeswitch-users] Problems subscribing to outbound socket events Message-ID: I've put some c++ test code together to let the outbound socket control the call, all works as expected, apart from the event subscription Sending myevents\n\n gives the channel events However sending event text all\n\n doesn't give me any events apart from the channel events. Anyone care to suggest what I might be doing wrong? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/b9e06398/attachment.html From gcd at i.ph Sat Jun 6 16:46:14 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 7 Jun 2009 07:46:14 +0800 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> Message-ID: <7d0bfd8c0906061646p4e5fd28eh741d6a57064de32e@mail.gmail.com> i'm using version build 13245M on an Intel D945GCLF2 Atom Dual-core mobo w/ 2GB ram. -nandy On Sat, Jun 6, 2009 at 11:02 AM, Brian West wrote: > You shouldn't be having problems... what version are you using? > /b > > On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: > > there 10 client seats so at max. 10 simultaneous calls. however, the number > of clients may be increased. > -nandy > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/42a56f51/attachment.html From gerry at pstn2.net Sat Jun 6 21:25:09 2009 From: gerry at pstn2.net (Gerry Hull) Date: Sun, 7 Jun 2009 00:25:09 -0400 Subject: [Freeswitch-users] I need a favor... Message-ID: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> OK, thanks to help on the list have my very cool FreeSwitch app running... Gotta love FS once you get over the learning hump! So, I build FS and got everything running smoothly on my Wndows development box. Great. Then I went to deploy it on a production server. As I figured, no copy-and-run here. I tried building the setup project but it's just not happening for me! Can someone out there build me the Windows MSI for build 13496 or later and provide a link to it? I'm in a bind here to get this up and running. If I can pry a few bux out of the boss, I hope to be a ClueCon and describe to application we have built with FreeSwitch. Regards, Gerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/200553b8/attachment.html From tayeb.meftah at gmail.com Sun Jun 7 01:08:32 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 07 Jun 2009 08:08:32 +0000 Subject: [Freeswitch-users] I need a favor... In-Reply-To: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> References: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> Message-ID: <4A2B7580.9020909@gmail.com> hello, welcome, i'm able to build a Installer for your Freeswitch please: i have a very small Internet connection (128KBPS) and FS Setup syse is 40MB or +... i give you only the setup project and you compile it... ok? thanks Gerry Hull wrote: > OK, thanks to help on the list have my very cool FreeSwitch app > running... Gotta love FS once you get over the learning hump! > > So, I build FS and got everything running smoothly on my Wndows > development box. Great. Then I went to deploy it on a production > server. As I figured, no copy-and-run here. > > I tried building the setup project but it's just not happening for me! > > Can someone out there build me the Windows MSI for build 13496 or > later and provide a link to it? I'm in a bind here to get this up > and running. > > If I can pry a few bux out of the boss, I hope to be a ClueCon and > describe to application we have built with FreeSwitch. > > Regards, > > Gerry > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/ff5419ad/attachment.html From anthony.minessale at gmail.com Sun Jun 7 07:32:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 7 Jun 2009 09:32:20 -0500 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <7d0bfd8c0906061646p4e5fd28eh741d6a57064de32e@mail.gmail.com> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> <7d0bfd8c0906061646p4e5fd28eh741d6a57064de32e@mail.gmail.com> Message-ID: <191c3a030906070732n677cb84k37941d026c8a401b@mail.gmail.com> Could you describe "latency"? not everyone uses it the same way. Maybe describe your exact problem. On Sat, Jun 6, 2009 at 6:46 PM, Nandy Dagondon wrote: > i'm using version build 13245M on an Intel D945GCLF2 Atom Dual-core mobo w/ > 2GB ram. > -nandy > > On Sat, Jun 6, 2009 at 11:02 AM, Brian West wrote: > >> You shouldn't be having problems... what version are you using? >> /b >> >> On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: >> >> there 10 client seats so at max. 10 simultaneous calls. however, the >> number of clients may be increased. >> -nandy >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/c09f5619/attachment.html From anthony.minessale at gmail.com Sun Jun 7 07:36:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 7 Jun 2009 09:36:15 -0500 Subject: [Freeswitch-users] Problems subscribing to outbound socket events In-Reply-To: References: Message-ID: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> once you send "myevents" you lock on to only that channel's events despite any other *events* command. you may want to use the filter feature on the channel's uuid instead. On Sat, Jun 6, 2009 at 12:39 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I?ve put some c++ test code together to let the outbound socket control > the call, all works as expected, apart from the event subscription > > > > Sending myevents\n\n gives the channel events > > > > However sending event text all\n\n doesn?t give me any events apart from the channel events. > > > > > > Anyone care to suggest what I might be doing wrong? > > > > Regards, > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/23f4d011/attachment-0001.html From anthony.minessale at gmail.com Sun Jun 7 07:37:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 7 Jun 2009 09:37:40 -0500 Subject: [Freeswitch-users] How to reject a call without answering In-Reply-To: <20090606042846.72320@gmx.net> References: <20090606033426.72340@gmx.net> <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> <20090606042846.72320@gmx.net> Message-ID: <191c3a030906070737p5c791316ub30845321bfdf7e9@mail.gmail.com> the 2nd one. On Fri, Jun 5, 2009 at 11:28 PM, Klaus Teller wrote: > It doesn't seem to work. I tried the following: > > api respond 9015430e-82cf-418c-bf4c-f3ac6e85caf2 503 > > > SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 > call-command: execute > execute-app-name: respond > execute-app-arg: 503 > > Is one of these what you meant? > > Klaus. > > -------- Original-Nachricht -------- > > Datum: Fri, 5 Jun 2009 22:45:29 -0500 > > Von: Brian West > > An: freeswitch-users at lists.freeswitch.org > > Betreff: Re: [Freeswitch-users] How to reject a call without answering > > > Try the respond app. > > > > /b > > > > On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: > > > > > Hi, > > > > > > Going through the socket api, how can i reject a call without having > > > to answer it first? > > > > > > I tried sending a hangup command with cause set either to NO_ANSWER > > > or NORMAL_CLEARING. In both cases, Freeswitch does create another > > > socket to deliver the very same call. > > > > > > More precisely, when a call comes in i send a connect command. Then > > > after some few seconds, i then send the following hangup command: > > > > > > SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f > > > call-command: hangup > > > hangup-cause: NO_ANSWER > > > > > > Thanks for any feedback. > > > > > > Klaus. > > > > > > -- > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > -- > GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 > Euro/mtl.! > http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/9abb451c/attachment.html From gerry at pstn2.net Sun Jun 7 09:31:19 2009 From: gerry at pstn2.net (Gerry Hull) Date: Sun, 7 Jun 2009 12:31:19 -0400 Subject: [Freeswitch-users] I need a favor... In-Reply-To: <4A2B7580.9020909@gmail.com> References: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> <4A2B7580.9020909@gmail.com> Message-ID: <98a86adf0906070931w71771cf3w83d12081f9eae568@mail.gmail.com> I have the installer project.. Can't get it to build. Can provIde ftp or will try ur project Thanks Gerry On 6/7/09, Meftah Tayeb wrote: > hello, > welcome, i'm able to build a Installer for your Freeswitch > please: > i have a very small Internet connection (128KBPS) and FS Setup syse is > 40MB or +... > i give you only the setup project and you compile it... ok? > thanks > Gerry Hull wrote: >> OK, thanks to help on the list have my very cool FreeSwitch app >> running... Gotta love FS once you get over the learning hump! >> >> So, I build FS and got everything running smoothly on my Wndows >> development box. Great. Then I went to deploy it on a production >> server. As I figured, no copy-and-run here. >> >> I tried building the setup project but it's just not happening for me! >> >> Can someone out there build me the Windows MSI for build 13496 or >> later and provide a link to it? I'm in a bind here to get this up >> and running. >> >> If I can pry a few bux out of the boss, I hope to be a ClueCon and >> describe to application we have built with FreeSwitch. >> >> Regards, >> >> Gerry >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > From brad.tuan at gmail.com Sun Jun 7 18:28:55 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Mon, 8 Jun 2009 09:28:55 +0800 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? Message-ID: As title, How to receive a call from another SIP proxy?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/8a57e2c9/attachment.html From brian at freeswitch.org Sun Jun 7 18:40:12 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 7 Jun 2009 20:40:12 -0500 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? In-Reply-To: References: Message-ID: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> Thats a rather broad question... can you tell us if you have to register to said proxy? /b On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote: > As title, > > How to receive a call from another SIP proxy?? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/fa16c68c/attachment.html From brad.tuan at gmail.com Sun Jun 7 19:00:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Mon, 8 Jun 2009 10:00:49 +0800 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? In-Reply-To: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> References: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> Message-ID: If myFS(203.64.xx.xx) want to receive a call from a SIP Proxy(163.28.xx.xx) In another word, User1(another SIP Proxy) want to call User2(FS), and User1 doesn't register to FS and User2 doesn't register to 163.28.xx.xx. How to set my FS?? 2009/6/8 Brian West > Thats a rather broad question... can you tell us if you have to register to > said proxy? > /b > > On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote: > > As title, > > How to receive a call from another SIP proxy?? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/502b86b0/attachment.html From jmesquita at gmail.com Sun Jun 7 20:13:41 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 8 Jun 2009 00:13:41 -0300 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? In-Reply-To: References: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> Message-ID: <5a8712120906072013r16915308v2a67ae53253ba916@mail.gmail.com> That is what the Sofia External profile is for! On the default config, it is set to receive SIP messages on port 5080. You should give some study on the default configs.... Take a look at the files (read all the nice comments) and follow it with the wiki. That would be the road to fortune. Later, jmesquita On Sun, Jun 7, 2009 at 11:00 PM, Brad Tuan wrote: > If myFS(203.64.xx.xx) want to receive a call from a SIP Proxy(163.28.xx.xx) > > In another word, User1(another SIP Proxy) want to call User2(FS), > > and User1 doesn't register to FS and User2 doesn't register to > 163.28.xx.xx. > > How to set my FS?? > 2009/6/8 Brian West > >> Thats a rather broad question... can you tell us if you have to register >> to said proxy? >> /b >> >> On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote: >> >> As title, >> >> How to receive a call from another SIP proxy?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/bd0a5cd7/attachment-0001.html From dujinfang at gmail.com Sun Jun 7 23:29:44 2009 From: dujinfang at gmail.com (seven) Date: Mon, 8 Jun 2009 14:29:44 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <7d0bfd8c0906052003w40a8fff4s8644cb9773e346b5@mail.gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> <7d0bfd8c0906052003w40a8fff4s8644cb9773e346b5@mail.gmail.com> Message-ID: <0E7D06F2-0964-4332-A5BF-440E723E47DF@gmail.com> On Jun 6, 2009, at 11:03 AM, Nandy Dagondon wrote: > dujinfang, > > hv u tried OSLEC? it's really reduced echo even on the cheapy X100P > card on *. oslec works w/ FS, too. Thanks, will try. :) > > > -nandy > > On Sat, Jun 6, 2009 at 10:11 AM, dujinfang > wrote: > > On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: >> >> >> On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: >> I'm using openzap analog with tone_detect, it works(conference not >> tested). however, according to the asterisk book, Kewlstart can >> detect the busy tone and disconnect the circuit. does anyone knows >> how to configure kewlstart with freeswitch/openzap? guess we don't >> need tone_detect then. >> >> Dujinfang, >> >> Your telco must support "kewlstart" signaling for this to be >> effective. The telco probably calls it something different, like >> "disconnect supervision" or "drop in loop current" or "battery >> reversal" or something like that. In any case, if the signaling is >> supported then you need to set up your zaptel.conf with the >> appropriate signaling type, which is either fxoks or fxsks. (I can >> never remember because zaptel does it backwards where if you have >> an FXO port then it uses FXS signaling but if you have an FXS port >> it uses FXO signaling. Stupidity, to be sure, so be aware of it.) >> > > 1) Don't know why but the similar zaptel.conf works on asterisk. I > guess tone_detect in FS is equivalent to busydetect=yes in > Asterisk(zapata.conf) . > > > zaptel.conf > # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) > fxsks=1 > fxsks=2 > fxsks=3 > fxsks=4 > > # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" > fxoks=5 > fxoks=6 > fxoks=7 > fxoks=8 > > # Global data > > loadzone = us > defaultzone = us > > > zapata.conf > > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=no > ;echotraining=800 > rxgain=0.0 > txgain=0.0 > group=0 > callgroup=1 > pickupgroup=1 > immediate=no > busydetect=yes > > I agree the FXO and FXS signaling is weird, why not they just match > the care name and reverse that internally? > > 2) Another essue is if I dial out from a FXO port from a local > extension(sip and zap), I can hear much echo on FreeSWITCH but not > much echo on asterisk. the zt.conf as below and I tried to change > the echo_cancel_level to 32 or 128 got no much difference. Is there > any equivalent configuration in FS like echocanccelwhenbridged=no in > asterisk? can I set busydetect and echocancelwhenbridged and other > options like this ? > > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > echo_cancel_level => 64 > > >> Find the sample zaptel.conf that comes with the zaptel package and >> search it for fxsks or fxoks and you'll see some notes on how to >> set it up for your analog trunks. >> >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/20ccf284/attachment.html From rdenert at tng.de Sun Jun 7 23:38:26 2009 From: rdenert at tng.de (Rudolf Denert) Date: Mon, 8 Jun 2009 08:38:26 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: Message-ID: <26244917.324011244443106580.JavaMail.root@zimbra.tng.de> @Brian West Hello, could I send you the traces from the wireshark directly to your e-mail address? The reason are the IP-addresses. I don't want to publish them to the whole world. Of course, you could send your answer to my question to freeswitch-users at lists.freeswitch.org. Best regards ----- Urspr?ngliche Mail ----- Von: "Brian West" An: freeswitch-users at lists.freeswitch.org Gesendet: Donnerstag, 4. Juni 2009 16:13:29 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Insufficient RTP stream Sorry but this type of trace is impossible to read. I want raw pcap if possible. /b On Jun 4, 2009, at 2:52 AM, Rudolf Denert wrote: Ok, here is the SIP trace. If you need more, just tell me and I will send them. The RTP trace you already have, haven't you? Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From rdenert at tng.de Sun Jun 7 23:42:19 2009 From: rdenert at tng.de (Rudolf Denert) Date: Mon, 8 Jun 2009 08:42:19 +0200 (CEST) Subject: [Freeswitch-users] DTMF Problems Message-ID: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> Hello! Is there a possibility to "detect" or "scan" which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? Greetz From durk.debeer at isp.solcon.nl Mon Jun 8 00:37:16 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Mon, 08 Jun 2009 09:37:16 +0200 Subject: [Freeswitch-users] Problem with attendant transfer Message-ID: Hello all, I have observed an issue on using Freeswitch and some SIP-phones. Ok the problem is this. Some phones, when attempting an attendant transfer, put the recipient of the transfer on hold. This results in Freeswitch starting to stream MOH music to the phone put on hold, if implemented. When now the original phone is pasing the transfer, Freeswitch is not going to process this transfer because the recipient end of it is on hold. It is however terminating the connections it has with the phone initiating the transfer. This means that the recipient of the transfer is never coming of hold again until it terminates the call. Is there a way to detect this behaviour, so I can get the recipient of hold before Freeswitch is processing the transfer?. Kind regards Durk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/f62c2247/attachment.html From brian at freeswitch.org Mon Jun 8 07:26:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 09:26:14 -0500 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> References: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> Message-ID: <93F60015-4D72-474C-BF5D-B5EBA7574D4C@freeswitch.org> This makes no sense.... Can you try to explain it more? I do attended transfers with sip phones every day without a single problem. Maybe i'm missing what you're talking about. /b On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote: > Hello all, > I have observed an issue on using Freeswitch and some SIP-phones. Ok > the problem is this. Some phones, when attempting an attendant > transfer, put the recipient of the transfer on hold. This results in > Freeswitch starting to stream MOH music to the phone put on hold, if > implemented. When now the original phone is pasing the transfer, > Freeswitch is not going to process this transfer because the > recipient end of it is on hold. It is however terminating the > connections it has with the phone initiating the transfer. This > means that the recipient of the transfer is never coming of hold > again until it terminates the call. > Is there a way to detect this behaviour, so I can get the recipient > of hold before Freeswitch is processing the transfer?. > Kind regards > Durk Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/6a0de4e6/attachment-0001.html From klaus.teller at gmx.net Mon Jun 8 07:27:37 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Mon, 08 Jun 2009 16:27:37 +0200 Subject: [Freeswitch-users] Taking long at startup Message-ID: <20090608142737.272910@gmx.net> Hi, Freeswitch is taking quiet some time to start. Is is normal these days? it didn't used to be the case few months ago. Is there anything i can turn off to start faster? Thanks, Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From intralanman at freeswitch.org Mon Jun 8 11:34:56 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 08 Jun 2009 14:34:56 -0400 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> References: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> Message-ID: <4A2D59D0.1020407@freeswitch.org> Durk de Beer wrote: > > Hello all, > > I have observed an issue on using Freeswitch and some SIP-phones. Ok > the problem is this. Some phones > which phones? -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/cb1ff4e2/attachment.html From brian at freeswitch.org Mon Jun 8 07:38:06 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 09:38:06 -0500 Subject: [Freeswitch-users] Taking long at startup In-Reply-To: <20090608142737.272910@gmx.net> References: <20090608142737.272910@gmx.net> Message-ID: <7E9EE8DC-9C8F-4D4A-80F2-36AFDCAD83C5@freeswitch.org> Update.. if it takes longer than 8 seconds start with -nonat /b On Jun 8, 2009, at 9:27 AM, Klaus Teller wrote: > Hi, > > Freeswitch is taking quiet some time to start. Is is normal these > days? it didn't used to be the case few months ago. Is there > anything i can turn off to start faster? > > Thanks, > Klaus. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/a4b06cc3/attachment.html From peter.olsson at visionutveckling.se Mon Jun 8 07:42:35 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 16:42:35 +0200 Subject: [Freeswitch-users] Taking long at startup In-Reply-To: <20090608142737.272910@gmx.net> References: <20090608142737.272910@gmx.net> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB88E4@cooper> Klaus, This is probably caused by the new nat features introduced in FreeSWITCH. You can start FS with -nonat to skip this detection. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Klaus Teller Skickat: den 8 juni 2009 16:28 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Taking long at startup Hi, Freeswitch is taking quiet some time to start. Is is normal these days? it didn't used to be the case few months ago. Is there anything i can turn off to start faster? Thanks, Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d22ba32931975679412! From kristian.kielhofner at gmail.com Mon Jun 8 08:08:59 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 8 Jun 2009 11:08:59 -0400 Subject: [Freeswitch-users] DTMF Problems In-Reply-To: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> References: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> Message-ID: <2d9149cd0906080808k792dd3d6i5af9282e4dce02ab@mail.gmail.com> Rudolf, I believe there is a snippet in the sample XML dialplan to detect the lack of telephone-event in the SDP and activate inband detection. You could use that for inspiration. On Mon, Jun 8, 2009 at 2:42 AM, Rudolf Denert wrote: > Hello! > > Is there a possibility to "detect" or "scan" which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? > > Greetz > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Jun 8 08:12:35 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 10:12:35 -0500 Subject: [Freeswitch-users] DTMF Problems In-Reply-To: <2d9149cd0906080808k792dd3d6i5af9282e4dce02ab@mail.gmail.com> References: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> <2d9149cd0906080808k792dd3d6i5af9282e4dce02ab@mail.gmail.com> Message-ID: I wrote that to demonstrate that exact situation but you still can't tell if they are inband or info :P /b On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote: > Rudolf, > > I believe there is a snippet in the sample XML dialplan to detect > the lack of telephone-event in the SDP and activate inband detection. > You could use that for inspiration. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/aa9873b4/attachment.html From rex.alex345 at yahoo.com Mon Jun 8 08:32:47 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 08:32:47 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> Message-ID: <1244475167805-3043665.post@n2.nabble.com> Hi Brian, I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 Addions made are, acl.conf.xml freeswitch.xml sip_profiles/external.xml under tag public.xml default.xml Still, Inbound is not hitting my FS console itself. Please assist where am I going wrong? Thanks, Rex Brian West wrote: > > > On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >> >> Hello, >> >> My public.xml configration is: >> >> >> >> >> >> > > $1 will not exist in this case because your regular expression doesn't > capture anything. So replace $1 with your target number or use > ^(123456)$ > > >> >> My default.xml configration is: >> >> >> >> >> >> >> >> > > Can you elaborate how you're registering with your provider? > > >> >> >> When I am trying to call 123456 from my mobile no. Not able to see any >> logging in FS console. Please assist where I am going wrong? Or do I >> require >> any extra modules to be installed? >> >> Thanks, >> Rex > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rex.alex345 at yahoo.com Mon Jun 8 08:35:50 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 08:35:50 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244475167805-3043665.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> <1244475167805-3043665.post@n2.nabble.com> Message-ID: <1244475350818-3043700.post@n2.nabble.com> Hi Brian, Missed.... sip_profiles/external.xml under tag Thanks, Rex Rex_Alex wrote: > > Hi Brian, > > I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > > Addions made are, > > acl.conf.xml > > > > > > freeswitch.xml > > > > sip_profiles/external.xml > > under tag > > public.xml > > > > > > > > default.xml > > > > > > > > > > Still, Inbound is not hitting my FS console itself. Please assist where am > I going wrong? > > Thanks, > Rex > > > > > > > > Brian West wrote: >> >> >> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >> >>> >>> Hello, >>> >>> My public.xml configration is: >>> >>> >>> >>> >>> >>> >> >> $1 will not exist in this case because your regular expression doesn't >> capture anything. So replace $1 with your target number or use >> ^(123456)$ >> >> >>> >>> My default.xml configration is: >>> >>> >>> >>> >>> >>> >>> >>> >> >> Can you elaborate how you're registering with your provider? >> >> >>> >>> >>> When I am trying to call 123456 from my mobile no. Not able to see any >>> logging in FS console. Please assist where I am going wrong? Or do I >>> require >>> any extra modules to be installed? >>> >>> Thanks, >>> Rex >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043700.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 08:44:19 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 17:44:19 +0200 Subject: [Freeswitch-users] Inbound using FS Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> Have you configured the sip profile to use the acl list you have created (Inbound_Test)? /Peter ----- Ursprungligt meddelande ----- Fr?n: Rex_Alex Skickat: den 8 juni 2009 17:40 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Inbound using FS Hi Brian, I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 Addions made are, acl.conf.xml freeswitch.xml sip_profiles/external.xml under tag public.xml default.xml Still, Inbound is not hitting my FS console itself. Please assist where am I going wrong? Thanks, Rex Brian West wrote: > > > On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >> >> Hello, >> >> My public.xml configration is: >> >> >> >> >> >> > > $1 will not exist in this case because your regular expression doesn't > capture anything. So replace $1 with your target number or use > ^(123456)$ > > >> >> My default.xml configration is: >> >> >> >> >> >> >> >> > > Can you elaborate how you're registering with your provider? > > >> >> >> When I am trying to call 123456 from my mobile no. Not able to see any >> logging in FS console. Please assist where I am going wrong? Or do I >> require >> any extra modules to be installed? >> >> Thanks, >> Rex > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d30e032931222027793! From rex.alex345 at yahoo.com Mon Jun 8 08:53:30 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 08:53:30 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> Message-ID: <1244476410874-3043804.post@n2.nabble.com> Hello Peter, Yes, I have added under tag in sip_profiles/external.xml Thanks, Rex Peter Olsson wrote: > > Have you configured the sip profile to use the acl list you have created > (Inbound_Test)? > > /Peter > > > ----- Ursprungligt meddelande ----- > Fr?n: Rex_Alex > Skickat: den 8 juni 2009 17:40 > Till: freeswitch-users at lists.freeswitch.org > > ?mne: Re: [Freeswitch-users] Inbound using FS > > > Hi Brian, > > I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > > Addions made are, > > acl.conf.xml > > > > > > freeswitch.xml > > > > sip_profiles/external.xml > > under tag > > public.xml > > > > > > > > default.xml > > > > > > data="sofia/internal/1007%1.1.1.1"/> > > > > Still, Inbound is not hitting my FS console itself. Please assist where am > I > going wrong? > > Thanks, > Rex > > > > > > > > Brian West wrote: >> >> >> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >> >>> >>> Hello, >>> >>> My public.xml configration is: >>> >>> >>> >>> >>> >>> >> >> $1 will not exist in this case because your regular expression doesn't >> capture anything. So replace $1 with your target number or use >> ^(123456)$ >> >> >>> >>> My default.xml configration is: >>> >>> >>> >>> >>> >>> >>> >>> >> >> Can you elaborate how you're registering with your provider? >> >> >>> >>> >>> When I am trying to call 123456 from my mobile no. Not able to see any >>> logging in FS console. Please assist where I am going wrong? Or do I >>> require >>> any extra modules to be installed? >>> >>> Thanks, >>> Rex >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d30e032931222027793! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 09:18:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 18:18:28 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244476410874-3043804.post@n2.nabble.com> Message-ID: I don't see what you've added. But I guess it's something like . Are you sure you're dialing into the external profile? It's on port 5080 by default, and the internal is on 5060. /Peter On 09-06-08 17.53, "Rex_Alex" wrote: Hello Peter, Yes, I have added under tag in sip_profiles/external.xml Thanks, Rex Peter Olsson wrote: > > Have you configured the sip profile to use the acl list you have created > (Inbound_Test)? > > /Peter > > > ----- Ursprungligt meddelande ----- > Fr?n: Rex_Alex > Skickat: den 8 juni 2009 17:40 > Till: freeswitch-users at lists.freeswitch.org > > ?mne: Re: [Freeswitch-users] Inbound using FS > > > Hi Brian, > > I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > > Addions made are, > > acl.conf.xml > > > > > > freeswitch.xml > > > > sip_profiles/external.xml > > under tag > > public.xml > > > > > > > > default.xml > > > > > > data="sofia/internal/1007%1.1.1.1"/> > > > > Still, Inbound is not hitting my FS console itself. Please assist where am > I > going wrong? > > Thanks, > Rex > > > > > > > > Brian West wrote: >> >> >> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >> >>> >>> Hello, >>> >>> My public.xml configration is: >>> >>> >>> >>> >>> >>> >> >> $1 will not exist in this case because your regular expression doesn't >> capture anything. So replace $1 with your target number or use >> ^(123456)$ >> >> >>> >>> My default.xml configration is: >>> >>> >>> >>> >>> >>> >>> >>> >> >> Can you elaborate how you're registering with your provider? >> >> >>> >>> >>> When I am trying to call 123456 from my mobile no. Not able to see any >>> logging in FS console. Please assist where I am going wrong? Or do I >>> require >>> any extra modules to be installed? >>> >>> Thanks, >>> Rex >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d354232931305258464! From rex.alex345 at yahoo.com Mon Jun 8 09:34:36 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 09:34:36 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> Message-ID: <1244478876050-3044052.post@n2.nabble.com> Hello, I am not sure about the profile which I am calling into. My scenario is this, I am trying to reach extn 1007 registered in FS server from my mobile through an inbound PRI connected to the audiocode with DID 123456. Thanks, Rex Peter Olsson wrote: > > I don't see what you've added. But I guess it's something like . > > Are you sure you're dialing into the external profile? It's on port 5080 > by default, and the internal is on 5060. > > /Peter > > > On 09-06-08 17.53, "Rex_Alex" wrote: > > > > Hello Peter, > > Yes, I have added > > > > under tag in sip_profiles/external.xml > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> Have you configured the sip profile to use the acl list you have created >> (Inbound_Test)? >> >> /Peter >> >> >> ----- Ursprungligt meddelande ----- >> Fr?n: Rex_Alex >> Skickat: den 8 juni 2009 17:40 >> Till: freeswitch-users at lists.freeswitch.org >> >> ?mne: Re: [Freeswitch-users] Inbound using FS >> >> >> Hi Brian, >> >> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >> >> Addions made are, >> >> acl.conf.xml >> >> >> >> >> >> freeswitch.xml >> >> >> >> sip_profiles/external.xml >> >> under tag >> >> public.xml >> >> >> >> >> >> >> >> default.xml >> >> >> >> >> >> > data="sofia/internal/1007%1.1.1.1"/> >> >> >> >> Still, Inbound is not hitting my FS console itself. Please assist where >> am >> I >> going wrong? >> >> Thanks, >> Rex >> >> >> >> >> >> >> >> Brian West wrote: >>> >>> >>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>> >>>> >>>> Hello, >>>> >>>> My public.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> $1 will not exist in this case because your regular expression doesn't >>> capture anything. So replace $1 with your target number or use >>> ^(123456)$ >>> >>> >>>> >>>> My default.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> Can you elaborate how you're registering with your provider? >>> >>> >>>> >>>> >>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>> logging in FS console. Please assist where I am going wrong? Or do I >>>> require >>>> any extra modules to be installed? >>>> >>>> Thanks, >>>> Rex >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d354232931305258464! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 09:50:57 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 18:50:57 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244478876050-3044052.post@n2.nabble.com> Message-ID: The PRI/SIP-box probably talks to the internal profile (I guess that they are on the same LAN). Try to add the inbound-acl to the internal profile as well. Also restart FS completely, just to be 100% sure that config is reloaded. //Peter On 09-06-08 18.34, "Rex_Alex" wrote: Hello, I am not sure about the profile which I am calling into. My scenario is this, I am trying to reach extn 1007 registered in FS server from my mobile through an inbound PRI connected to the audiocode with DID 123456. Thanks, Rex Peter Olsson wrote: > > I don't see what you've added. But I guess it's something like . > > Are you sure you're dialing into the external profile? It's on port 5080 > by default, and the internal is on 5060. > > /Peter > > > On 09-06-08 17.53, "Rex_Alex" wrote: > > > > Hello Peter, > > Yes, I have added > > > > under tag in sip_profiles/external.xml > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> Have you configured the sip profile to use the acl list you have created >> (Inbound_Test)? >> >> /Peter >> >> >> ----- Ursprungligt meddelande ----- >> Fr?n: Rex_Alex >> Skickat: den 8 juni 2009 17:40 >> Till: freeswitch-users at lists.freeswitch.org >> >> ?mne: Re: [Freeswitch-users] Inbound using FS >> >> >> Hi Brian, >> >> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >> >> Addions made are, >> >> acl.conf.xml >> >> >> >> >> >> freeswitch.xml >> >> >> >> sip_profiles/external.xml >> >> under tag >> >> public.xml >> >> >> >> >> >> >> >> default.xml >> >> >> >> >> >> > data="sofia/internal/1007%1.1.1.1"/> >> >> >> >> Still, Inbound is not hitting my FS console itself. Please assist where >> am >> I >> going wrong? >> >> Thanks, >> Rex >> >> >> >> >> >> >> >> Brian West wrote: >>> >>> >>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>> >>>> >>>> Hello, >>>> >>>> My public.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> $1 will not exist in this case because your regular expression doesn't >>> capture anything. So replace $1 with your target number or use >>> ^(123456)$ >>> >>> >>>> >>>> My default.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> Can you elaborate how you're registering with your provider? >>> >>> >>>> >>>> >>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>> logging in FS console. Please assist where I am going wrong? Or do I >>>> require >>>> any extra modules to be installed? >>>> >>>> Thanks, >>>> Rex >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d3f8432938250412368! From rex.alex345 at yahoo.com Mon Jun 8 10:04:36 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 10:04:36 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> Message-ID: <1244480676920-3044219.post@n2.nabble.com> Hello, Yes you are right. they are the on the same LAN. Inboun-acl added in internal profile as well and restarted the FS completely. But no luck.. Please help us to resolve the same.. Thanks, Rex Peter Olsson wrote: > > The PRI/SIP-box probably talks to the internal profile (I guess that they > are on the same LAN). Try to add the inbound-acl to the internal profile > as well. Also restart FS completely, just to be 100% sure that config is > reloaded. > > //Peter > > > On 09-06-08 18.34, "Rex_Alex" wrote: > > > > Hello, > > I am not sure about the profile which I am calling into. > > My scenario is this, I am trying to reach extn 1007 registered in FS > server > from my mobile through an inbound PRI connected to the audiocode with DID > 123456. > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> I don't see what you've added. But I guess it's something like . >> >> Are you sure you're dialing into the external profile? It's on port 5080 >> by default, and the internal is on 5060. >> >> /Peter >> >> >> On 09-06-08 17.53, "Rex_Alex" wrote: >> >> >> >> Hello Peter, >> >> Yes, I have added >> >> >> >> under tag in sip_profiles/external.xml >> >> Thanks, >> Rex >> >> >> >> Peter Olsson wrote: >>> >>> Have you configured the sip profile to use the acl list you have created >>> (Inbound_Test)? >>> >>> /Peter >>> >>> >>> ----- Ursprungligt meddelande ----- >>> Fr?n: Rex_Alex >>> Skickat: den 8 juni 2009 17:40 >>> Till: freeswitch-users at lists.freeswitch.org >>> >>> ?mne: Re: [Freeswitch-users] Inbound using FS >>> >>> >>> Hi Brian, >>> >>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >>> >>> Addions made are, >>> >>> acl.conf.xml >>> >>> >>> >>> >>> >>> freeswitch.xml >>> >>> >>> >>> sip_profiles/external.xml >>> >>> under tag >>> >>> public.xml >>> >>> >>> >>> >>> >>> >>> >>> default.xml >>> >>> >>> >>> >>> >>> >> data="sofia/internal/1007%1.1.1.1"/> >>> >>> >>> >>> Still, Inbound is not hitting my FS console itself. Please assist where >>> am >>> I >>> going wrong? >>> >>> Thanks, >>> Rex >>> >>> >>> >>> >>> >>> >>> >>> Brian West wrote: >>>> >>>> >>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>>> >>>>> >>>>> Hello, >>>>> >>>>> My public.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> $1 will not exist in this case because your regular expression doesn't >>>> capture anything. So replace $1 with your target number or use >>>> ^(123456)$ >>>> >>>> >>>>> >>>>> My default.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> Can you elaborate how you're registering with your provider? >>>> >>>> >>>>> >>>>> >>>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>>> logging in FS console. Please assist where I am going wrong? Or do I >>>>> require >>>>> any extra modules to be installed? >>>>> >>>>> Thanks, >>>>> Rex >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d3f8432938250412368! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 10:22:43 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 19:22:43 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244480676920-3044219.post@n2.nabble.com> Message-ID: Are you able to see anything at all in the console/log? I'm starting to doubt that the call even gets into the FS box... :) Try enabling more logs (console loglevel debug), and try again. On 09-06-08 19.04, "Rex_Alex" wrote: Hello, Yes you are right. they are the on the same LAN. Inboun-acl added in internal profile as well and restarted the FS completely. But no luck.. Please help us to resolve the same.. Thanks, Rex Peter Olsson wrote: > > The PRI/SIP-box probably talks to the internal profile (I guess that they > are on the same LAN). Try to add the inbound-acl to the internal profile > as well. Also restart FS completely, just to be 100% sure that config is > reloaded. > > //Peter > > > On 09-06-08 18.34, "Rex_Alex" wrote: > > > > Hello, > > I am not sure about the profile which I am calling into. > > My scenario is this, I am trying to reach extn 1007 registered in FS > server > from my mobile through an inbound PRI connected to the audiocode with DID > 123456. > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> I don't see what you've added. But I guess it's something like . >> >> Are you sure you're dialing into the external profile? It's on port 5080 >> by default, and the internal is on 5060. >> >> /Peter >> >> >> On 09-06-08 17.53, "Rex_Alex" wrote: >> >> >> >> Hello Peter, >> >> Yes, I have added >> >> >> >> under tag in sip_profiles/external.xml >> >> Thanks, >> Rex >> >> >> >> Peter Olsson wrote: >>> >>> Have you configured the sip profile to use the acl list you have created >>> (Inbound_Test)? >>> >>> /Peter >>> >>> >>> ----- Ursprungligt meddelande ----- >>> Fr?n: Rex_Alex >>> Skickat: den 8 juni 2009 17:40 >>> Till: freeswitch-users at lists.freeswitch.org >>> >>> ?mne: Re: [Freeswitch-users] Inbound using FS >>> >>> >>> Hi Brian, >>> >>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >>> >>> Addions made are, >>> >>> acl.conf.xml >>> >>> >>> >>> >>> >>> freeswitch.xml >>> >>> >>> >>> sip_profiles/external.xml >>> >>> under tag >>> >>> public.xml >>> >>> >>> >>> >>> >>> >>> >>> default.xml >>> >>> >>> >>> >>> >>> >> data="sofia/internal/1007%1.1.1.1"/> >>> >>> >>> >>> Still, Inbound is not hitting my FS console itself. Please assist where >>> am >>> I >>> going wrong? >>> >>> Thanks, >>> Rex >>> >>> >>> >>> >>> >>> >>> >>> Brian West wrote: >>>> >>>> >>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>>> >>>>> >>>>> Hello, >>>>> >>>>> My public.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> $1 will not exist in this case because your regular expression doesn't >>>> capture anything. So replace $1 with your target number or use >>>> ^(123456)$ >>>> >>>> >>>>> >>>>> My default.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> Can you elaborate how you're registering with your provider? >>>> >>>> >>>>> >>>>> >>>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>>> logging in FS console. Please assist where I am going wrong? Or do I >>>>> require >>>>> any extra modules to be installed? >>>>> >>>>> Thanks, >>>>> Rex >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d45dc32931463593608! From timb0311 at hotmail.com Mon Jun 8 11:31:44 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 8 Jun 2009 14:31:44 -0400 Subject: [Freeswitch-users] Routing 911 calls In-Reply-To: References: Message-ID: If I have 1 inbound DID and "client software" that calls this DID from multiple locations (states, city, etc), how would I go about placing and routing a 911 / emergency calls? Keep in mind each client does have their own existing phone line (DID) with 911/e911 attached to that DID. Could I route by setting outbound caller id and number to the incoming, and then place the outbound 911 call? Tim _________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/ade2bb75/attachment.html From brian at freeswitch.org Mon Jun 8 11:43:37 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 13:43:37 -0500 Subject: [Freeswitch-users] Routing 911 calls In-Reply-To: References: Message-ID: Not sure that'll meet the legal requirements. /b On Jun 8, 2009, at 1:31 PM, Tim B wrote: > Could I route by setting outbound caller id and number to the > incoming, and then place the outbound 911 call? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/ae224788/attachment.html From msc at freeswitch.org Mon Jun 8 12:03:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 8 Jun 2009 12:03:55 -0700 Subject: [Freeswitch-users] Routing 911 calls In-Reply-To: References: Message-ID: <87f2f3b90906081203w595f4f56k19579b7459e72ab3@mail.gmail.com> On Mon, Jun 8, 2009 at 11:31 AM, Tim B wrote: > If I have 1 inbound DID and "client software" that calls this DID from > multiple locations (states, city, etc), how would I go about placing and > routing a 911 / emergency calls? Keep in mind each client does have their > own existing phone line (DID) with 911/e911 attached to that DID. > > Could I route by setting outbound caller id and number to the incoming, and > then place the outbound 911 call? > > You can try it, and then make a test 911 call and tell the operator that you are testing a new phone system install. The 911 op can then verify the information he/she sees on the screen with where you are. If you're in New York when you call but the 911 op sees "Jerkwater, Alabama" then you know it doesn't work... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/599843e3/attachment-0001.html From brian at freeswitch.org Mon Jun 8 12:08:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 14:08:40 -0500 Subject: [Freeswitch-users] Sounds for 1.0.4 release and zRTP sound files. Message-ID: I have a deal from GM Voices for this order at a major discount but its still $650 USD, If you wish to donate to this order please let me know... brian at freeswitch.org is my paypal. Thanks, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/5049a2d2/attachment.html From c_cav_01 at yahoo.com Mon Jun 8 13:18:49 2009 From: c_cav_01 at yahoo.com (Chris) Date: Mon, 8 Jun 2009 13:18:49 -0700 (PDT) Subject: [Freeswitch-users] Routing 911 calls Message-ID: <796626.25975.qm@web55103.mail.re4.yahoo.com> Looking for a reasonable DID/e911 provider.? Any suggestions. p.s.? Sorry to hijack the thread.? --- On Mon, 6/8/09, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] Routing 911 calls To: freeswitch-users at lists.freeswitch.org Date: Monday, June 8, 2009, 1:03 PM On Mon, Jun 8, 2009 at 11:31 AM, Tim B wrote: If I have 1 inbound DID and "client software"?that calls this DID from multiple locations (states, city, etc), how would I go about placing and routing?a 911 / emergency?calls?? Keep in mind each client does have their own existing phone line (DID)?with 911/e911 attached to that DID. ? Could I route by setting outbound caller id and number to the incoming, and then place the outbound 911 call? ? You can try it, and then make a test 911 call and tell the operator that you are testing a new phone system install. The 911 op can then verify the information he/she sees on the screen with where you are. If you're in New York when you call but the 911 op sees "Jerkwater, Alabama" then you know it doesn't work... -MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/5a82a92a/attachment.html From msc at freeswitch.org Mon Jun 8 14:31:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 8 Jun 2009 14:31:23 -0700 Subject: [Freeswitch-users] Problems subscribing to outbound socket events In-Reply-To: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> References: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> Message-ID: <87f2f3b90906081431n38500f1fi35ee7e5d314733d7@mail.gmail.com> On Sun, Jun 7, 2009 at 7:36 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > once you send "myevents" you lock on to only that channel's events despite > any other *events* command. > you may want to use the filter feature on the channel's uuid instead. > > FYI, I've added this information to the event socket wiki page. There wasn't an entry on the filter command so I added one with some simple examples. Hope this helps people in the future. http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/8413e8e2/attachment.html From jmesquita at gmail.com Mon Jun 8 15:33:41 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 8 Jun 2009 19:33:41 -0300 Subject: [Freeswitch-users] Problems subscribing to outbound socket events In-Reply-To: <87f2f3b90906081431n38500f1fi35ee7e5d314733d7@mail.gmail.com> References: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> <87f2f3b90906081431n38500f1fi35ee7e5d314733d7@mail.gmail.com> Message-ID: <5a8712120906081533p198d3c2bq2a758f774a0f986e@mail.gmail.com> MC is tha man... jmesquita On Mon, Jun 8, 2009 at 6:31 PM, Michael Collins wrote: > > > On Sun, Jun 7, 2009 at 7:36 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> once you send "myevents" you lock on to only that channel's events despite >> any other *events* command. >> you may want to use the filter feature on the channel's uuid instead. >> >> > FYI, I've added this information to the event socket wiki page. There > wasn't an entry on the filter command so I added one with some simple > examples. Hope this helps people in the future. > > http://wiki.freeswitch.org/wiki/Mod_event_socket#filter > > -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/a55f50d3/attachment.html From john at feith.com Mon Jun 8 17:40:22 2009 From: john at feith.com (John Wehle) Date: Mon, 8 Jun 2009 20:40:22 -0400 (EDT) Subject: [Freeswitch-users] Caller id when doing transfers Message-ID: <200906090040.n590eMHr004215@jwlab.FEITH.COM> Consider the following sequence: 1) Outside caller (OC) calls Ext 1001 Caller id shows OC 2) Ext 1001 transfers the call to Ext 1002 In some cases we want the caller id to shows OC, in other cases we want the caller id to show Ext 1001. It appears from some limited testing that the original caller id is always shown when the call is transfered. Is there some way to have the person making the transfer show up as the caller id? Our application is I want to setup an extension (*5) which automatically places calls into a fifo corresponding to the Extension number of the person transferring the call. This will provide park capability similar to that of our old System 25 PBX. I'll then setup an extension (*8) which picks up the call at the front of the fifo "fifo". -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From jim at evolutiontel.net Mon Jun 8 17:44:00 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 9 Jun 2009 10:44:00 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: <51DECC6F-E822-4459-BF47-970576041825@freeswitch.org> References: <51DECC6F-E822-4459-BF47-970576041825@freeswitch.org> Message-ID: Gents, Thanks for your input on this.....much appreciated. Changing the codec from G729a to G729 before the call hits FS did the trick. Calls are now going through correctly. Cheers, Jim On Sat, Jun 6, 2009 at 12:32 PM, Brian West wrote: > G729a is 100% INVALID in the sdp on codec 18. > /b > On Jun 5, 2009, at 9:25 PM, Jim Burke wrote: > > Noticed there is a list of mime types associated with FS and G729a was not > listed, does this have anything to do with the root cause? > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From larclap at yahoo.com Mon Jun 8 19:24:46 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 8 Jun 2009 19:24:46 -0700 Subject: [Freeswitch-users] Can't hear outbound calls Message-ID: <005001c9e8a9$76c5c780$64515680$@com> I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/869885cf/attachment-0001.html From brian at freeswitch.org Mon Jun 8 19:38:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 21:38:39 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <005001c9e8a9$76c5c780$64515680$@com> References: <005001c9e8a9$76c5c780$64515680$@com> Message-ID: <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> pastebin your profile config and the output of global_getvar /b On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: > I had a working FS installation which I messed up by doing a fresh > install. I tried to integrate all my custom changes, but I?m sure I > screwed something up. > > The symptom is on an outbound call, sometimes I can hear ringing, > other times I cannot. Finally I can see FS connects via a softphone, > but I hear only silence. The other side of the conversation hears > static. > > I pasted a siptrace of the external profile. The Contact, Via and > SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 > > FS exists on a LAN behind a NAT firewall along with all its clients. > There is a SwitchVox system which predates the FS. I had to use an > external sip = 5090 for FS. Also I think I had to use a different > WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) > than the one FS came up with (xxx.xxx.xxx.82), but I can?t figure > out where I set this address. > > I would appreciate any help. Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/1e9006e9/attachment.html From larclap at yahoo.com Mon Jun 8 19:57:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 8 Jun 2009 19:57:45 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> Message-ID: <006101c9e8ae$11f03c00$35d0b400$@com> http://pastebin.freeswitch.org/9319 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 08, 2009 7:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls pastebin your profile config and the output of global_getvar /b On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/9e2f9a3a/attachment.html From jason at jasonjgw.net Mon Jun 8 20:30:51 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 9 Jun 2009 13:30:51 +1000 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <005001c9e8a9$76c5c780$64515680$@com> References: <005001c9e8a9$76c5c780$64515680$@com> Message-ID: <20090609033051.GA28848@jdc.jasonjgw.net> Lars Zeb wrote: > I had a working FS installation which I messed up by doing a fresh install. > I tried to integrate all my custom changes, but I'm sure I screwed something > up. Git is an excellent tool for keeping track of FreeSWITCH configuration changes. The history of my configuration is maintained in a git repository under /opt/freeswitch/conf - git simply creates a .git subdirectory to store all of the revisions as they are committed. Git revert and git stash have been very useful at times, not to mention git reset --hard. Since Git is used for Linux kernel development, it should be available from most recent Linux distributions, and it can probably be compiled for other Unix-like environments as well. From saigop at gmail.com Mon Jun 8 21:47:21 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Tue, 9 Jun 2009 10:17:21 +0530 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1244480676920-3044219.post@n2.nabble.com> Message-ID: <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> Hi Rex, You need to allow your acl in internal.xml like the one, Change the internal-network according to your configuration you allowed in acl.conf.xml. I have tested with audiocode with PRI line its working fine. On Mon, Jun 8, 2009 at 10:52 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Are you able to see anything at all in the console/log? > > I'm starting to doubt that the call even gets into the FS box... :) > > Try enabling more logs (console loglevel debug), and try again. > > > On 09-06-08 19.04, "Rex_Alex" wrote: > > > > > > Hello, > > Yes you are right. they are the on the same LAN. Inboun-acl added > in internal profile as well and restarted the FS completely. But no luck.. > Please help us to resolve the same.. > > Thanks, > Rex > > > Peter Olsson wrote: > > > > The PRI/SIP-box probably talks to the internal profile (I guess that they > > are on the same LAN). Try to add the inbound-acl to the internal profile > > as well. Also restart FS completely, just to be 100% sure that config is > > reloaded. > > > > //Peter > > > > > > On 09-06-08 18.34, "Rex_Alex" wrote: > > > > > > > > Hello, > > > > I am not sure about the profile which I am calling into. > > > > My scenario is this, I am trying to reach extn 1007 registered in FS > > server > > from my mobile through an inbound PRI connected to the audiocode with DID > > 123456. > > > > Thanks, > > Rex > > > > > > > > Peter Olsson wrote: > >> > >> I don't see what you've added. But I guess it's something like . > >> > >> Are you sure you're dialing into the external profile? It's on port 5080 > >> by default, and the internal is on 5060. > >> > >> /Peter > >> > >> > >> On 09-06-08 17.53, "Rex_Alex" wrote: > >> > >> > >> > >> Hello Peter, > >> > >> Yes, I have added > >> > >> > >> > >> under tag in sip_profiles/external.xml > >> > >> Thanks, > >> Rex > >> > >> > >> > >> Peter Olsson wrote: > >>> > >>> Have you configured the sip profile to use the acl list you have > created > >>> (Inbound_Test)? > >>> > >>> /Peter > >>> > >>> > >>> ----- Ursprungligt meddelande ----- > >>> Fr?n: Rex_Alex > >>> Skickat: den 8 juni 2009 17:40 > >>> Till: freeswitch-users at lists.freeswitch.org > >>> > >>> ?mne: Re: [Freeswitch-users] Inbound using FS > >>> > >>> > >>> Hi Brian, > >>> > >>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > >>> > >>> Addions made are, > >>> > >>> acl.conf.xml > >>> > >>> > >>> > >>> > >>> > >>> freeswitch.xml > >>> > >>> > >>> > >>> sip_profiles/external.xml > >>> > >>> under tag > >>> > >>> public.xml > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> default.xml > >>> > >>> > >>> > >>> > >>> > >>> >>> data="sofia/internal/1007%1.1.1.1"/> > >>> > >>> > >>> > >>> Still, Inbound is not hitting my FS console itself. Please assist where > >>> am > >>> I > >>> going wrong? > >>> > >>> Thanks, > >>> Rex > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> Brian West wrote: > >>>> > >>>> > >>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >>>> > >>>>> > >>>>> Hello, > >>>>> > >>>>> My public.xml configration is: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>> > >>>> $1 will not exist in this case because your regular expression doesn't > >>>> capture anything. So replace $1 with your target number or use > >>>> ^(123456)$ > >>>> > >>>> > >>>>> > >>>>> My default.xml configration is: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> data="sofia/sip/1001%freeswitchip"/> > >>>>> > >>>>> > >>>> > >>>> Can you elaborate how you're registering with your provider? > >>>> > >>>> > >>>>> > >>>>> > >>>>> When I am trying to call 123456 from my mobile no. Not able to see > any > >>>>> logging in FS console. Please assist where I am going wrong? Or do I > >>>>> require > >>>>> any extra modules to be installed? > >>>>> > >>>>> Thanks, > >>>>> Rex > >>>> > >>>> Brian West > >>>> brian at freeswitch.org > >>>> > >>>> -- Meet us at ClueCon! http://www.cluecon.com > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> > >>> -- > >>> View this message in context: > >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html > >>> Sent from the freeswitch-users mailing list archive at Nabble.com. > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -- > >> View this message in context: > >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html > >> Sent from the freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d45dc32931463593608! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/152e72bf/attachment-0001.html From talk2ram at gmail.com Tue Jun 9 00:02:41 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 12:32:41 +0530 Subject: [Freeswitch-users] mod_niible install problem Message-ID: Hi i have downloaded latest SVN and trying to make install i get the following error I googled for the same but there no information on this error how can i resolve this problem Ram making install mod_nibblebill Compiling mod_nibblebill.c... Compiling mod_nibblebill.c ... mod_nibblebill.c: In function ?get_balance?: mod_nibblebill.c:368: error: ?balance? undeclared (first use in this function) mod_nibblebill.c:368: error: (Each undeclared identifier is reported only once mod_nibblebill.c:368: error: for each function it appears in.) make[5]: *** [mod_nibblebill.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_nibblebill-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/89610d47/attachment.html From talk2ram at gmail.com Tue Jun 9 00:29:30 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 12:59:30 +0530 Subject: [Freeswitch-users] I need a favor... In-Reply-To: <4A2B7580.9020909@gmail.com> References: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> <4A2B7580.9020909@gmail.com> Message-ID: On Sun, Jun 7, 2009 at 1:38 PM, Meftah Tayeb wrote: > hello, > welcome, i'm able to build a Installer for your Freeswitch > please: > i have a very small Internet connection (128KBPS) and FS Setup syse is 40MB > or +... > i give you only the setup project and you compile it... ok? > thanks > is this linux based or windows ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/32c9179d/attachment.html From yivzhenko at mksat.net Tue Jun 9 01:26:03 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 9 Jun 2009 11:26:03 +0300 Subject: [Freeswitch-users] mod_nibblebill not set variable nibble_total_billed Message-ID: <200906091126.03556.yivzhenko@mksat.net> Some time ago mod_nibblebill was set variable nibble_total_billed after hangup. But after last few updates of module this variable is no more sets. Somebody else have this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9cf2cf56/attachment.html From talk2ram at gmail.com Tue Jun 9 01:36:11 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 14:06:11 +0530 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: On Sun, May 31, 2009 at 8:39 PM, bakko wrote: > If you understand spanish please look at: > > http://www.freeswitch.es/node/55 > > Hi i have followed this URL iam using mysql, when i run the following command iam getting error what iam doing wrong ? python manage.py syncdb Traceback (most recent call last): File "manage.py", line 30, in from django.core.management import execute_manager ImportError: No module named django.core.management Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/5fdaa665/attachment.html From durk.debeer at isp.solcon.nl Tue Jun 9 01:43:36 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Tue, 9 Jun 2009 10:43:36 +0200 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: References: Message-ID: Ok I've recieved an error message so if this message is being send a second time my deepest apologies for it. Hello Brian, I observed the problem by a Siemens Gigaset DE380 IPR and A Cisco 7960 and 7965. What happens is this, a call is coming in on Freeswitch and is being bridged to lets say the Siemens on extension 100. Now the person accepting the call on extension 100 wants to transfer the call to an other extension lets say 200. The normal scenario would be extension 100 puts the original call on hold, Freeswitch streams moh to the original call, extension 100 dials 200, the having a conversation, 100 transfers the original call too 200. Now here the problem begins. Normally extension 100 would send a refer sip message to freeswitch, who would then connect the original call to extension 200. The Siemens and Cisco phones do not send a refer sip message first. What the do is putting extension 200 on hold by means of an send only sip message. When Freeswitch is receiving this it streams moh to extension 200. After this the phones are sending the transfer by means of a sip refer message. When Freeswitch is receiving this it can't perform this transfer, that of original call to extension 200, because extension 100 has put extension 200 on receive only and extension 200 is receiving the moh. Resulting in an original call receiving moh and an extension 200 receiving moh. When this situation arises there's no way in connecting these to together. So what I need is a way to detect that there is an transfer by means of an sip refer message to a extension that has being put on hold. If so I need to get freeswitch to break this hold and transfer the original call to this extension. I hope that this will make the problem a little bit clearer. Kind regards Durk > This makes no sense.... Can you try to explain it more? I do attended > transfers with sip phones every day without a single problem. Maybe > i'm missing what you're talking about. > /b > On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote: >> Hello all, >> I have observed an issue on using Freeswitch and some SIP-phones. Ok >> the problem is this. Some phones, when attempting an attendant >> transfer, put the recipient of the transfer on hold. This results in >> Freeswitch starting to stream MOH music to the phone put on hold, if >> implemented. When now the original phone is pasing the transfer, >> Freeswitch is not going to process this transfer because the >> recipient end of it is on hold. It is however terminating the >> connections it has with the phone initiating the transfer. This >> means that the recipient of the transfer is never coming of hold >> again until it terminates the call. >> Is there a way to detect this behaviour, so I can get the recipient >> of hold before Freeswitch is processing the transfer?. >> Kind regards >> Durk > Brian West > brian at freeswitch.org From talk2ram at gmail.com Tue Jun 9 02:00:50 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 14:30:50 +0530 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: On Tue, Jun 9, 2009 at 2:06 PM, ram wrote: > > > On Sun, May 31, 2009 at 8:39 PM, bakko wrote: > >> If you understand spanish please look at: >> >> http://www.freeswitch.es/node/55 >> >> > > Hi > > i have followed this URL > > iam using mysql, > > when i run the following command iam getting error > > > what iam doing wrong ? > > python manage.py syncdb > Traceback (most recent call last): > File "manage.py", line 30, in > from django.core.management import execute_manager > ImportError: No module named django.core.management > > iam able to fix the above problem as per mentioned in the below URL http://forum.webfaction.com/viewtopic.php?id=324 now iam not able to start Freeswitch ./freeswitch 2009-06-09 01:57:45.826955 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-09 01:57:45.829038 [DEBUG] switch_event.c:552 Create event dispatch thread 0 Cannot Initialize [[error near line 972]: missing >] Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/b711a883/attachment.html From mikael at bjerkeland.com Tue Jun 9 02:02:36 2009 From: mikael at bjerkeland.com (Mikael Aleksander Bjerkeland) Date: Tue, 09 Jun 2009 11:02:36 +0200 Subject: [Freeswitch-users] video transcoding In-Reply-To: <1244046640.28699.63.camel@localhost.localdomain> References: <1244046640.28699.63.camel@localhost.localdomain> Message-ID: <1244538156.4732.11.camel@mikael-xpsm1530> Hi, El mi?, 03-06-2009 a las 18:30 +0200, Francois Delawarde escribi?: > Hello, > > I'm interested in being able to do video transcoding mainly for > bridging 3G mobile and sip networks, and maybe later on some > conferencing with FS. > > Are video codecs planned to be added to FS even in a far future? Are > there copyright/patent problems with common video codecs (H.263 / > H.264) or with libraries (ffmpeg) that would prevent any of that from > happening? Yes, there are copyright/patent problems. The x264 library should be able to do transcoding but its license is not compatible with FS and some of the x264 developers are not interested in releasing it under a dual license. > > Meanwhile, would it be feasible to do some video transcoding using > external software (vlc?) with socket connections from-to FS? A guy I talked to in #x264 on FreeNode claims to be doing this, but not in a VoIP scenario. If you don't want to reinvent the wheel you probably have to create a wrapper for your socket interface to interface with x264. Please share your progress if you do so. If all else fails you could have a look at these to bridge the video call to PRI http://www.mirial.com/products/PSE_3G_Gateway.html http://www.radvision.com/Products/3GProductsApplications/SCOPIA3GVideoGateway/ And no, I do not know if they work :-) > > Thanks, > Fran?ois. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raffaele.p.guidi at gmail.com Tue Jun 9 02:14:58 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 9 Jun 2009 11:14:58 +0200 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: A side question: has anyone succesfully installed wikipbx on windows? On Tue, Jun 9, 2009 at 11:00, ram wrote: > > > On Tue, Jun 9, 2009 at 2:06 PM, ram wrote: > >> >> >> On Sun, May 31, 2009 at 8:39 PM, bakko wrote: >> >>> If you understand spanish please look at: >>> >>> http://www.freeswitch.es/node/55 >>> >>> >> >> Hi >> >> i have followed this URL >> >> iam using mysql, >> >> when i run the following command iam getting error >> >> >> what iam doing wrong ? >> >> python manage.py syncdb >> Traceback (most recent call last): >> File "manage.py", line 30, in >> from django.core.management import execute_manager >> ImportError: No module named django.core.management >> >> > > iam able to fix the above problem as per mentioned in the below URL > > http://forum.webfaction.com/viewtopic.php?id=324 > now iam not able to start Freeswitch > > ./freeswitch > 2009-06-09 01:57:45.826955 [INFO] switch_event.c:564 Activate Eventing > Engine. > 2009-06-09 01:57:45.829038 [DEBUG] switch_event.c:552 Create event dispatch > thread 0 > Cannot Initialize [[error near line 972]: missing >] > > > Ram > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/6f1664e8/attachment-0001.html From asannucci at gmail.com Tue Jun 9 02:41:35 2009 From: asannucci at gmail.com (bakko) Date: Tue, 9 Jun 2009 11:41:35 +0200 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com><9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com><99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: Look at /usr/local/freeswitch/log/freeswitch.xml.fsxml Line 972 This is a sintaxys error in the configuration. Chao -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/f8e08312/attachment.html From talk2ram at gmail.com Tue Jun 9 02:46:32 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 15:16:32 +0530 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: On Tue, Jun 9, 2009 at 3:11 PM, bakko wrote: > Look at /usr/local/freeswitch/log/freeswitch.xml.fsxml > > Line 972 > > This is a sintaxys error in the configuration. > > it was manual mistake when i edited that file fixed problem Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8f2c7677/attachment.html From durk.debeer at isp.solcon.nl Tue Jun 9 02:55:24 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Tue, 9 Jun 2009 11:55:24 +0200 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: References: Message-ID: <1D1333C8C667434989D600F21BFFFC5E@solcon.local> Hallo Ray Siemens type Gigaset DE380 IPR, Cisco type 7965 and 7960. Also tested Grandstream type 2010 and Linksys type 921 no problem with these phones. I've downloaded a new firmware for the Siemens but wasn't able to test it jet. Durk > Durk de Beer wrote: >> >> Hello all, >> >> I have observed an issue on using Freeswitch and some SIP-phones. Ok >> the problem is this. Some phones >> > which phones? > -Ray From klaus.teller at gmx.net Tue Jun 9 05:17:10 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 09 Jun 2009 14:17:10 +0200 Subject: [Freeswitch-users] Reject call without answering Message-ID: <20090609121710.157260@gmx.net> Hi, I'm still looking for a way to reject a call without answering. I've tried various things without solution. >From the socket interface i tried: SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 call-command: execute execute-app-name: respond execute-app-arg: 503 >From Javascript, i tried each of the followings: session.hangup(); session.reject(); session.execute("respond", data="503"); In the two first cases as well as in the socket interface case, it seemed as if Freeswitch sent a kill signal to the VoiP provider. But that isn't enough apparently to cancel the call. In the third case, i get the message: Session is not active. Any other suggestion? Thanks, Klaus. -- GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 Euro/mtl.! http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a From rdenert at tng.de Tue Jun 9 05:56:49 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 9 Jun 2009 14:56:49 +0200 (CEST) Subject: [Freeswitch-users] DTMF Problems In-Reply-To: <5346021.351571244551867400.JavaMail.root@zimbra.tng.de> Message-ID: <11031925.351641244552209113.JavaMail.root@zimbra.tng.de> Hello, I can give the all-clear! It was my mistake ( ...and it was a silly one :-/ ) I had to applications that interfere each other. They are: I don't know why I skip that in my dialplan!!! http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Yes! A small mistake with a huge effect. But thanks for all you help. :-) Greetz ----- Urspr?ngliche Mail ----- Von: "Brian West" An: freeswitch-users at lists.freeswitch.org Gesendet: Montag, 8. Juni 2009 17:12:35 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] DTMF Problems I wrote that to demonstrate that exact situation but you still can't tell if they are inband or info :P /b On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote: Rudolf, I believe there is a snippet in the sample XML dialplan to detect the lack of telephone-event in the SDP and activate inband detection. You could use that for inspiration. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Jun 9 06:45:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 08:45:02 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <006101c9e8ae$11f03c00$35d0b400$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> Message-ID: You have a upnp device handing out 0.0.0.0 as the gateway address ... I'll patch that shortly to disable that. /b On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote: > http://pastebin.freeswitch.org/9319 > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Monday, June 08, 2009 7:39 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can't hear outbound calls > > pastebin your profile config and the output of global_getvar > > /b > > On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: > > > I had a working FS installation which I messed up by doing a fresh > install. I tried to integrate all my custom changes, but I?m sure I > screwed something up. > > The symptom is on an outbound call, sometimes I can hear ringing, > other times I cannot. Finally I can see FS connects via a softphone, > but I hear only silence. The other side of the conversation hears > static. > > I pasted a siptrace of the external profile. The Contact, Via and > SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 > > FS exists on a LAN behind a NAT firewall along with all its clients. > There is a SwitchVox system which predates the FS. I had to use an > external sip = 5090 for FS. Also I think I had to use a different > WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) > than the one FS came up with (xxx.xxx.xxx.82), but I can?t figure > out where I set this address. > > I would appreciate any help. Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/6e8853c6/attachment-0001.html From rex.alex345 at yahoo.com Tue Jun 9 06:46:06 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 9 Jun 2009 06:46:06 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> <1244480676920-3044219.post@n2.nabble.com> Message-ID: <1244555166651-3049491.post@n2.nabble.com> Incoming calls are not even hitting my FS box console. Peter Olsson wrote: > > Are you able to see anything at all in the console/log? > > I'm starting to doubt that the call even gets into the FS box... :) > > Try enabling more logs (console loglevel debug), and try again. > > > On 09-06-08 19.04, "Rex_Alex" wrote: > > > > > > Hello, > > Yes you are right. they are the on the same LAN. Inboun-acl added > in internal profile as well and restarted the FS completely. But no luck.. > Please help us to resolve the same.. > > Thanks, > Rex > > > Peter Olsson wrote: >> >> The PRI/SIP-box probably talks to the internal profile (I guess that they >> are on the same LAN). Try to add the inbound-acl to the internal profile >> as well. Also restart FS completely, just to be 100% sure that config is >> reloaded. >> >> //Peter >> >> >> On 09-06-08 18.34, "Rex_Alex" wrote: >> >> >> >> Hello, >> >> I am not sure about the profile which I am calling into. >> >> My scenario is this, I am trying to reach extn 1007 registered in FS >> server >> from my mobile through an inbound PRI connected to the audiocode with DID >> 123456. >> >> Thanks, >> Rex >> >> >> >> Peter Olsson wrote: >>> >>> I don't see what you've added. But I guess it's something like . >>> >>> Are you sure you're dialing into the external profile? It's on port 5080 >>> by default, and the internal is on 5060. >>> >>> /Peter >>> >>> >>> On 09-06-08 17.53, "Rex_Alex" wrote: >>> >>> >>> >>> Hello Peter, >>> >>> Yes, I have added >>> >>> >>> >>> under tag in sip_profiles/external.xml >>> >>> Thanks, >>> Rex >>> >>> >>> >>> Peter Olsson wrote: >>>> >>>> Have you configured the sip profile to use the acl list you have >>>> created >>>> (Inbound_Test)? >>>> >>>> /Peter >>>> >>>> >>>> ----- Ursprungligt meddelande ----- >>>> Fr?n: Rex_Alex >>>> Skickat: den 8 juni 2009 17:40 >>>> Till: freeswitch-users at lists.freeswitch.org >>>> >>>> ?mne: Re: [Freeswitch-users] Inbound using FS >>>> >>>> >>>> Hi Brian, >>>> >>>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >>>> >>>> Addions made are, >>>> >>>> acl.conf.xml >>>> >>>> >>>> >>>> >>>> >>>> freeswitch.xml >>>> >>>> >>>> >>>> sip_profiles/external.xml >>>> >>>> under tag >>>> >>>> public.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> default.xml >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/internal/1007%1.1.1.1"/> >>>> >>>> >>>> >>>> Still, Inbound is not hitting my FS console itself. Please assist where >>>> am >>>> I >>>> going wrong? >>>> >>>> Thanks, >>>> Rex >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Brian West wrote: >>>>> >>>>> >>>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>>>> >>>>>> >>>>>> Hello, >>>>>> >>>>>> My public.xml configration is: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> $1 will not exist in this case because your regular expression doesn't >>>>> capture anything. So replace $1 with your target number or use >>>>> ^(123456)$ >>>>> >>>>> >>>>>> >>>>>> My default.xml configration is: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/sip/1001%freeswitchip"/> >>>>>> >>>>>> >>>>> >>>>> Can you elaborate how you're registering with your provider? >>>>> >>>>> >>>>>> >>>>>> >>>>>> When I am trying to call 123456 from my mobile no. Not able to see >>>>>> any >>>>>> logging in FS console. Please assist where I am going wrong? Or do I >>>>>> require >>>>>> any extra modules to be installed? >>>>>> >>>>>> Thanks, >>>>>> Rex >>>>> >>>>> Brian West >>>>> brian at freeswitch.org >>>>> >>>>> -- Meet us at ClueCon! http://www.cluecon.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d45dc32931463593608! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3049491.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mrene_lists at avgs.ca Tue Jun 9 06:50:30 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 9 Jun 2009 09:50:30 -0400 Subject: [Freeswitch-users] mod_niible install problem In-Reply-To: References: Message-ID: <32EACEF7-594E-4332-8E6A-4230E76C79FA@avgs.ca> Hi, Install unixodbc-dev and run configure again. Math On 9-Jun-09, at 3:02 AM, ram wrote: > > Hi > > i have downloaded latest SVN > > and trying to make install > > i get the following error > > I googled for the same > but there no information on this error > > > how can i resolve this problem > > Ram > > making install mod_nibblebill > Compiling mod_nibblebill.c... > Compiling mod_nibblebill.c ... > mod_nibblebill.c: In function ?get_balance?: > mod_nibblebill.c:368: error: ?balance? undeclared (first use in this > function) > mod_nibblebill.c:368: error: (Each undeclared identifier is reported > only once > mod_nibblebill.c:368: error: for each function it appears in.) > make[5]: *** [mod_nibblebill.lo] Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_nibblebill-install] Error 1 > make[2]: *** [install-recursive] Error 1 > Making install in build > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rex.alex345 at yahoo.com Tue Jun 9 06:55:19 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 9 Jun 2009 06:55:19 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> <1244480676920-3044219.post@n2.nabble.com> <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> Message-ID: <1244555719062-3049537.post@n2.nabble.com> Hello, Below are some changes I have made, Post me if any additions required... acl.conf.xml freeswitch.xml sip_profiles/internal.xml < param name="apply-inbound-acl" value="inbound_ac" /> under tag public.xml default.xml Thanks Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3049537.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Tue Jun 9 07:26:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 9 Jun 2009 16:26:28 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244555719062-3049537.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> <1244480676920-3044219.post@n2.nabble.com> <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> <1244555719062-3049537.post@n2.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB8A7C@cooper> If you don't even see it when debug logging is enabled, there is something wrong in the other end. About the IP's. I guess you're just faking IP's in these email,s or are you using 2.2.2.2 and 1.1.1.1 for real? Cause in that case you're in trouble. I just wanted to make sure... :) /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Rex_Alex Skickat: den 9 juni 2009 15:55 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Inbound using FS Hello, Below are some changes I have made, Post me if any additions required... acl.conf.xml freeswitch.xml sip_profiles/internal.xml < param name="apply-inbound-acl" value="inbound_ac" /> under tag public.xml default.xml Thanks Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3049537.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2e6b2232934656730255! From brian at freeswitch.org Tue Jun 9 07:41:14 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 09:41:14 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <006101c9e8ae$11f03c00$35d0b400$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> Message-ID: <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> update to 13719, rupa did a patch that fixed this.. also find that printer that gives out the 0.0.0.0 addr and turn off upnp :P /b On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote: > http://pastebin.freeswitch.org/9319 > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Monday, June 08, 2009 7:39 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can't hear outbound calls > > pastebin your profile config and the output of global_getvar > > /b > > On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: > > > I had a working FS installation which I messed up by doing a fresh > install. I tried to integrate all my custom changes, but I?m sure I > screwed something up. > > The symptom is on an outbound call, sometimes I can hear ringing, > other times I cannot. Finally I can see FS connects via a softphone, > but I hear only silence. The other side of the conversation hears > static. > > I pasted a siptrace of the external profile. The Contact, Via and > SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 > > FS exists on a LAN behind a NAT firewall along with all its clients. > There is a SwitchVox system which predates the FS. I had to use an > external sip = 5090 for FS. Also I think I had to use a different > WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) > than the one FS came up with (xxx.xxx.xxx.82), but I can?t figure > out where I set this address. > > I would appreciate any help. Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/1fdda4fa/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Jun 9 08:09:40 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 9 Jun 2009 16:09:40 +0100 Subject: [Freeswitch-users] mod vmd and lua - Solved Message-ID: I finally got around to looking at why mod vmd didn't appear to run when using LUA. Turned out that the example in the wiki was wrong. It should have been session:execute("vmd","start"); And not session:execute("vmd"); I've updated the wiki Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8f8932b2/attachment.html From msc at freeswitch.org Tue Jun 9 08:44:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 08:44:24 -0700 Subject: [Freeswitch-users] ClueCon 2009 Information - Roommates Message-ID: <87f2f3b90906090844u4797759ew170f470cd19e0330@mail.gmail.com> FYI, We've had several people inquire about finding someone with whom to share a room for ClueCon. Please keep in mind that we do have a minimum number of rooms we need to book with the Wyndham. However, we would hate for someone not to attend simply because they could not afford a hotel room. To that end I would like to ask for a volunteer from the community to be the go-to person for helping roommates to get connected. Perhaps we could get more total rooms booked by helping those with rooming needs and who might not otherwise be able to come to ClueCon this year. Please email me off list if you are able to help. Thanks! -Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/6bc2a7fb/attachment.html From msc at freeswitch.org Tue Jun 9 08:45:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 08:45:25 -0700 Subject: [Freeswitch-users] mod vmd and lua - Solved In-Reply-To: References: Message-ID: <87f2f3b90906090845n5e00e8d4r1436d55aa69f6a5@mail.gmail.com> > > > > I?ve updated the wiki > > > > You're a gentleman and a scholar! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/78a565ca/attachment.html From larclap at yahoo.com Tue Jun 9 08:54:41 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 08:54:41 -0700 Subject: [Freeswitch-users] Error making FS configure Message-ID: <012001c9e91a$99c4c1c0$cd4e4540$@com> I tried to make FS current with the following command, which got errors in the ./configure step. Is it OK to proceed to make install? make clean && svn up && ./configure configure: creating ./config.status config.status: creating Makefile config.status: creating libedit.pc config.status: creating src/Makefile config.status: creating doc/Makefile config.status: creating examples/Makefile config.status: creating config.h config.status: executing depfiles commands configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/76dbc731/attachment.html From gmaruzz at celliax.org Tue Jun 9 09:00:31 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 9 Jun 2009 18:00:31 +0200 Subject: [Freeswitch-users] broken compilation on windows? Message-ID: <7b197bef0906090900v6052e150vcff69406135d5645@mail.gmail.com> Hi all, I cannot compile on Windows the current svn, 13722. The first error it gives is: freeswitch\libs\pcre\pcre_internal.h(368) : fatal error C1189: #error : LINK_SIZE must be either 2, 3, or 4 then it fails 81 projects (42 succeeded), because no freeswitchcore.lib (obviously) I tried both the Freeswitch.2008.sln and the freeswitch.express.2008.sln, I'm using VC Express 2008. -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From krice at freeswitch.org Tue Jun 9 09:01:28 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 09 Jun 2009 11:01:28 -0500 Subject: [Freeswitch-users] Error making FS configure In-Reply-To: <012001c9e91a$99c4c1c0$cd4e4540$@com> Message-ID: Re-bootstrap if you just updated several fixes to the build system and an update to the pcre lab were recently commited... If you follow the ?trunk mailing everyone tries to tag commits that require this in their commit messages From: Lars Zeb Reply-To: Date: Tue, 9 Jun 2009 08:54:41 -0700 To: Subject: [Freeswitch-users] Error making FS configure I tried to make FS current with the following command, which got errors in the ../configure step. Is it OK to proceed to make install? make clean && svn up && ./configure configure: creating ./config.status config.status: creating Makefile config.status: creating libedit.pc config.status: creating src/Makefile config.status: creating doc/Makefile config.status: creating examples/Makefile config.status: creating config.h config.status: executing depfiles commands configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/02ecc57b/attachment.html From larclap at yahoo.com Tue Jun 9 09:03:20 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 09:03:20 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> Message-ID: <012501c9e91b$cf654880$6e2fd980$@com> Brian, I'm curious, how can you tell that a printer is giving out the 0.0.0.0 addr? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 7:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls update to 13719, rupa did a patch that fixed this.. also find that printer that gives out the 0.0.0.0 addr and turn off upnp :P /b On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote: http://pastebin.freeswitch.org/9319 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 08, 2009 7:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls pastebin your profile config and the output of global_getvar /b On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/292ed15a/attachment-0001.html From mike at jerris.com Tue Jun 9 09:06:03 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 9 Jun 2009 12:06:03 -0400 Subject: [Freeswitch-users] Error making FS configure In-Reply-To: References: Message-ID: <4384608B-DEEE-4ADD-B532-5756077FBF6C@jerris.com> I updated the pcre lib last night, and it is trying to add files that are already there in your working copy. You can try doing: rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make current Mike On Jun 9, 2009, at 12:01 PM, Ken Rice wrote: > Re-bootstrap if you just updated several fixes to the build system > and an update to the pcre lab were recently commited... If you > follow the ?trunk mailing everyone tries to tag commits that require > this in their commit messages > > > From: Lars Zeb > Reply-To: > Date: Tue, 9 Jun 2009 08:54:41 -0700 > To: > Subject: [Freeswitch-users] Error making FS configure > > I tried to make FS current with the following command, which got > errors in the ../configure step. Is it OK to proceed to make install? > > make clean && svn up && ./configure > configure: creating ./config.status > config.status: creating Makefile > config.status: creating libedit.pc > config.status: creating src/Makefile > config.status: creating doc/Makefile > config.status: creating examples/Makefile > config.status: creating config.h > config.status: executing depfiles commands > configure: configuring in libs/pcre > configure: running /bin/sh './configure.gnu' --prefix=/usr/local/ > freeswitch --cache-file=/dev/null --srcdir=. > configure: error: cannot find sources (pcre.h.in) in . > configure: error: /bin/sh './configure.gnu' failed for libs/pcre > > Thanks, Lars > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/17edbe46/attachment.html From brian at freeswitch.org Tue Jun 9 09:06:09 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 11:06:09 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <012501c9e91b$cf654880$6e2fd980$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> Message-ID: Because rupa on IRC is having the same problem.. Check the error message I print now and the device url will be printed thanks to rupa's patch. /b On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: > Brian, I?m curious, how can you tell that a printer is giving out > the 0.0.0.0 addr? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/e5f27091/attachment.html From eman at chabotel.com Tue Jun 9 09:14:08 2009 From: eman at chabotel.com (freeswitch list) Date: Tue, 9 Jun 2009 12:14:08 -0400 Subject: [Freeswitch-users] Few questions Message-ID: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> 1) How do you setup a gateway where the provider doesn't require a user name and password? For freeswitch gateways username and password are mandatory. 2) Is there anyway to have xml_curl send the password for directory entry requests. 3) What is the best way to monitor for failed sip register attempts? 4) Is there a way to increase the volume of voicemail message sent by email? They are fine in freeswitch but when sent by email as mp3 (mod_shout) they are really low. 5) Is there any future plans to make the voicemail module more customizable? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/fac939e2/attachment.html From msc at freeswitch.org Tue Jun 9 09:36:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 09:36:20 -0700 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> Message-ID: <87f2f3b90906090936t21d0919k717e553fa3e18a6d@mail.gmail.com> On Tue, Jun 9, 2009 at 9:14 AM, freeswitch list wrote: > 1) How do you setup a gateway where the provider doesn't require a user > name and password? For freeswitch gateways username and password are > mandatory. > Find this line in example.xml: Uncomment that line in your gateway and it won't attempt to register. Put a bogus username and password in those fields to keep FS from complaining and you're all set. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/73ef0f74/attachment.html From brian at freeswitch.org Tue Jun 9 09:43:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 11:43:16 -0500 Subject: [Freeswitch-users] Few questions In-Reply-To: <87f2f3b90906090936t21d0919k717e553fa3e18a6d@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <87f2f3b90906090936t21d0919k717e553fa3e18a6d@mail.gmail.com> Message-ID: <6F01F1FD-09F0-4563-A439-68A404089CB4@freeswitch.org> But the bigger issue is if the far end doesn't require register and won't 401/407 you and does auth via IP you don't need a gateway. /b On Jun 9, 2009, at 11:36 AM, Michael Collins wrote: > > > On Tue, Jun 9, 2009 at 9:14 AM, freeswitch list > wrote: > 1) How do you setup a gateway where the provider doesn't require a > user name and password? For freeswitch gateways username and > password are mandatory. > > Find this line in example.xml: > > > Uncomment that line in your gateway and it won't attempt to > register. Put a bogus username and password in those fields to keep > FS from complaining and you're all set. > -MC Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/786a471c/attachment.html From rupa at rupa.com Tue Jun 9 09:44:04 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 11:44:04 -0500 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> Message-ID: 2009/6/9 freeswitch list > 1) How do you setup a gateway where the provider doesn't require a user > name and password? For freeswitch gateways username and password are > mandatory. > Use the param: > > 2) Is there anyway to have xml_curl send the password for directory entry > requests. > dunno > > 3) What is the best way to monitor for failed sip register attempts? > You should get an event over event sockets on registration failure. You can also watch the log file for registration failure. Finally, "sofia status" will give you the current status of all gateways. You should also enable ping if you want to fail faster after the gateway becomes unreachable. > > 4) Is there a way to increase the volume of voicemail message sent by > email? They are fine in freeswitch but when sent by email as mp3 > (mod_shout) they are really low. > Look in shout.conf.xml -- there is a volume and outscale parameter. I haven't looked at the source to see exactly what they are for, but that should get you started. > > 5) Is there any future plans to make the voicemail module more > customizable? > If you can code -- patches welcome. Otherwise, you can file a request on jira with what you want. Add a bounty to encourage it to get done sooner rather than later. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/526d748a/attachment-0001.html From gmaruzz at celliax.org Tue Jun 9 09:45:59 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 9 Jun 2009 18:45:59 +0200 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems Message-ID: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From dujinfang at gmail.com Tue Jun 9 09:48:52 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 00:48:52 +0800 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> Message-ID: <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> > 2) Is there anyway to have xml_curl send the password for directory > entry requests. > try this: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7b8cc95c/attachment.html From larclap at yahoo.com Tue Jun 9 10:00:04 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 10:00:04 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> Message-ID: <015a01c9e923$bf458020$3dd08060$@com> Thanks, Brian and Mike J and Ken R and Jason W. Outbound calls are now working OK. Brian, I don't know where to look for the 0.0.0.0 addr error message. I checked the log/freeswitch.log but did not recognize anything. I also noticed that nat_public_addr is not longer displayed in the global_getvar command. How is this value set? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 9:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls Because rupa on IRC is having the same problem.. Check the error message I print now and the device url will be printed thanks to rupa's patch. /b On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: Brian, I'm curious, how can you tell that a printer is giving out the 0.0.0.0 addr? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/de40e2ee/attachment.html From brian at freeswitch.org Tue Jun 9 10:06:29 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 12:06:29 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015a01c9e923$bf458020$3dd08060$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> Message-ID: <6497F66C-4E87-412E-B222-7B6DEA789392@freeswitch.org> because 0.0.0.0 causes it to disable upnp cuz its invalid. /b On Jun 9, 2009, at 12:00 PM, Lars Zeb wrote: > Thanks, Brian and Mike J and Ken R and Jason W. > > Outbound calls are now working OK. > > Brian, I don?t know where to look for the 0.0.0.0 addr error > message. I checked the log/freeswitch.log but did not recognize > anything. > > I also noticed that nat_public_addr is not longer displayed in the > global_getvar command. How is this value set? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/507a9dbd/attachment.html From teqx at yahoo.com Tue Jun 9 10:14:49 2009 From: teqx at yahoo.com (zigurds) Date: Tue, 9 Jun 2009 10:14:49 -0700 (PDT) Subject: [Freeswitch-users] bug in channel event other leg parameters? Message-ID: <23947374.post@talk.nabble.com> Hi, all. While expecting channel events that are fired in call lifetime, have encountered that for lot of channel events Other-Leg-Username points to user that belongs to channel event for what is fired. For example, let's say I'm calling from 101 to 102, when receiving events CHANNEL_UNBRIDGE, CHANNEL_EXECUTE_COMPLETE, CHANNEL_HANGUP, if channel name is Channel-Name: sofia/Test/101%40XXXX, I'm getting also Other-Leg-Username: 101. I was expected that there should be 102 or I'm misunderstanding something? Some other information in Other-Leg too releates to user 101. By the way, does there exist more detailed information about call events flow and what event attributes mean. Seems that http://wiki.freeswitch.org/wiki/Event_list is very uncomplete... Thanks, Zigurds -- View this message in context: http://www.nabble.com/bug-in-channel-event-other-leg-parameters--tp23947374p23947374.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From eman at chabotel.com Tue Jun 9 10:26:53 2009 From: eman at chabotel.com (freeswitch list) Date: Tue, 9 Jun 2009 13:26:53 -0400 Subject: [Freeswitch-users] Few questions In-Reply-To: <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> Message-ID: <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> Awesome. Thank you so much guys. One last question. How would I forward a call in my dialplan to my cell phone? I tried but the caller id changes to the gateway callerid. On Tue, Jun 9, 2009 at 12:48 PM, dujinfang wrote: > 2) Is there anyway to have xml_curl send the password for directory entry > requests. > > try this: > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/17b5eb8e/attachment-0001.html From rupa at rupa.com Tue Jun 9 11:15:47 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 13:15:47 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015a01c9e923$bf458020$3dd08060$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> Message-ID: If we get 0.0.0.0 as a public address, upnp support is disabled since we're getting the info from a gateway that can't route us to the internet. It is broken, so we don't trust it. The message you should see in the logs is something like: 2009-06-08 13:51:44.587812 [ERR] switch_nat.c:126 uPNP Device (url: http://192.168.1.2:4444/wipconn) returned an invalid external address of 0.0.0.0. Disabling uPNP 2009-06-08 13:51:44.587812 [INFO] switch_nat.c:380 No PMP or UPnP NAT detected! You'll probably not see this if you start fs in the background and then connect with fs_cli. So, look in your log files for it. The url will give you an idea as to which device is sending you invalid info. In my case it is a dlink router setup as a access point but still was implementing upnp (bad). I was able to disable upnp on that router. My printer (Epson Artisan 800) also participates in upnp, but it doesn't respond to the internet gateway stuff, so it was not the source of a problem for address discovery. It is causing me other issues but that is another story for another day for code that isn't committed yet. 2009/6/9 Lars Zeb > Thanks, Brian and Mike J and Ken R and Jason W. > > > > Outbound calls are now working OK. > > > > Brian, I don?t know where to look for the 0.0.0.0 addr error message. I > checked the log/freeswitch.log but did not recognize anything. > > > > I also noticed that nat_public_addr is not longer displayed in the > global_getvar command. How is this value set? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Tuesday, June 09, 2009 9:06 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > Because rupa on IRC is having the same problem.. Check the error message I > print now and the device url will be printed thanks to rupa's patch. > > > > /b > > > > On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: > > > > Brian, I?m curious, how can you tell that a printer is giving out the > 0.0.0.0 addr? > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/19d8e8f1/attachment.html From eman at chabotel.com Tue Jun 9 11:16:26 2009 From: eman at chabotel.com (freeswitch list) Date: Tue, 9 Jun 2009 14:16:26 -0400 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> Message-ID: <164a9ab00906091116gfef6046vec2362ef2d51c53a@mail.gmail.com> I couldn't get the gateway to work with register=false. I took the hint from Brian and scraped the gateway idea and put this in my dialplan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/123c05a5/attachment.html From freeswitch-users at digitaldan.com Tue Jun 9 13:28:09 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Tue, 9 Jun 2009 14:28:09 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <25563227.12121244578744238.JavaMail.daniel@radio> Message-ID: <32308728.12141244579283168.JavaMail.daniel@radio> Hi guys, I'm noticing that when recording calls (using session.recordFile in both lua and javascript) the call is disconnecting after 12 hours (exactly 12 hours). I'm still looking into our cisco gateways to see if they are doing the call clearing, but wanted to know if there were any timers on calls or specically on recording calls. I'm using svn rev 13471 on debian, the calls originate from a cisco 53xx media gateway and are using sip / g.711 ulaw. Thanks. Dan- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/ab9ce5fd/attachment.html From brian at freeswitch.org Tue Jun 9 13:39:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 15:39:30 -0500 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <32308728.12141244579283168.JavaMail.daniel@radio> References: <32308728.12141244579283168.JavaMail.daniel@radio> Message-ID: how big is the file? /b On Jun 9, 2009, at 3:28 PM, freeswitch-users at digitaldan.com wrote: > Hi guys, > > I'm noticing that when recording calls (using session.recordFile in > both lua and javascript) the call is disconnecting after 12 hours > (exactly 12 hours). I'm still looking into our cisco gateways to > see if they are doing the call clearing, but wanted to know if > there were any timers on calls or specically on recording calls. > I'm using svn rev 13471 on debian, the calls originate from a cisco > 53xx media gateway and are using sip / g.711 ulaw. > > Thanks. > Dan- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/d3b3d609/attachment.html From freeswitch-users at digitaldan.com Tue Jun 9 13:57:09 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 9 Jun 2009 14:57:09 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: Message-ID: <32268694.12161244581025121.JavaMail.daniel@radio> 330M (345723426 bytes) D- ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 9, 2009 2:39:30 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording time limit? how big is the file? /b On Jun 9, 2009, at 3:28 PM, freeswitch-users at digitaldan.com wrote: Hi guys, I'm noticing that when recording calls (using session.recordFile in both lua and javascript) the call is disconnecting after 12 hours (exactly 12 hours). I'm still looking into our cisco gateways to see if they are doing the call clearing, but wanted to know if there were any timers on calls or specically on recording calls. I'm using svn rev 13471 on debian, the calls originate from a cisco 53xx media gateway and are using sip / g.711 ulaw. Thanks. Dan- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/800fc6bb/attachment-0001.html From larclap at yahoo.com Tue Jun 9 14:26:00 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 14:26:00 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> Message-ID: <020001c9e948$e2cca7b0$a865f710$@com> Rupa, Thanks for the detailed response. After upgrading from 13639 to 13732, I see no log errors. I am accessing Freeswitch vi fs_cli, but I did look in log/freeswitch.log. Certainly I see nothing that looks like your ERR below. I too have a dlink router. I will look at its configuration and see if upnp is enabled. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 11:16 AM To: freeswitch-users Subject: Re: [Freeswitch-users] Can't hear outbound calls If we get 0.0.0.0 as a public address, upnp support is disabled since we're getting the info from a gateway that can't route us to the internet. It is broken, so we don't trust it. The message you should see in the logs is something like: 2009-06-08 13:51:44.587812 [ERR] switch_nat.c:126 uPNP Device (url: http://192.168.1.2:4444/wipconn) returned an invalid external address of 0.0.0.0. Disabling uPNP 2009-06-08 13:51:44.587812 [INFO] switch_nat.c:380 No PMP or UPnP NAT detected! You'll probably not see this if you start fs in the background and then connect with fs_cli. So, look in your log files for it. The url will give you an idea as to which device is sending you invalid info. In my case it is a dlink router setup as a access point but still was implementing upnp (bad). I was able to disable upnp on that router. My printer (Epson Artisan 800) also participates in upnp, but it doesn't respond to the internet gateway stuff, so it was not the source of a problem for address discovery. It is causing me other issues but that is another story for another day for code that isn't committed yet. 2009/6/9 Lars Zeb Thanks, Brian and Mike J and Ken R and Jason W. Outbound calls are now working OK. Brian, I don't know where to look for the 0.0.0.0 addr error message. I checked the log/freeswitch.log but did not recognize anything. I also noticed that nat_public_addr is not longer displayed in the global_getvar command. How is this value set? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 9:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls Because rupa on IRC is having the same problem.. Check the error message I print now and the device url will be printed thanks to rupa's patch. /b On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: Brian, I'm curious, how can you tell that a printer is giving out the 0.0.0.0 addr? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/81946ee9/attachment.html From brian at freeswitch.org Tue Jun 9 14:32:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 16:32:30 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <020001c9e948$e2cca7b0$a865f710$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> Message-ID: You have to start freeswitch without -nc to see it. Only happens during start up. /b On Jun 9, 2009, at 4:26 PM, Lars Zeb wrote: > Rupa, > > Thanks for the detailed response. After upgrading from 13639 to > 13732, I see no log errors. I am accessing Freeswitch vi fs_cli, but > I did look in log/freeswitch.log. Certainly I see nothing that looks > like your ERR below. > > I too have a dlink router. I will look at its configuration and see > if upnp is enabled. > > Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9cadd1a0/attachment.html From brian at freeswitch.org Tue Jun 9 14:36:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 16:36:26 -0500 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <32268694.12161244581025121.JavaMail.daniel@radio> References: <32268694.12161244581025121.JavaMail.daniel@radio> Message-ID: What format are you recording in ? And what is the hangup cause? Happen to have a sip trace? /b On Jun 9, 2009, at 3:57 PM, Dan wrote: > 330M (345723426 bytes) > > D- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/4012318d/attachment.html From freeswitch-users at digitaldan.com Tue Jun 9 14:52:19 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 9 Jun 2009 15:52:19 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: Message-ID: <9615666.12391244584331944.JavaMail.daniel@radio> The freeswitch log show this before it hits my hangup hook in lua. I don't have a pcap trace of the sip messaging, I can try that tonight, it would definitely show where the hangup is coming from. I'm recording to a local disk as a .ul file, so a headerless ulaw format. 2009-06-08 21:25:44 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/external/5555551212 at 192.168.3.21 entering state [terminated][200] 2009-06-08 21:25:44 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/external/5555551212 at 192.168.3.21 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-08 21:25:44 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/external/5555551212 at 192.168.3.21 [KILL] 2009-06-08 21:25:44 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/external/5555551212 at 192.168.3.21 [BREAK] 2009-06-08 21:25:44 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read_codec() Restore original codec. 2009-06-08 21:25:44 [NOTICE] switch_cpp.cpp:1122 console_log() hangupHook: 272423 status: hangup Thanks D- ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 9, 2009 3:36:26 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording time limit? What format are you recording in ? And what is the hangup cause? Happen to have a sip trace? /b On Jun 9, 2009, at 3:57 PM, Dan wrote: 330M (345723426 bytes) D- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/929b03b8/attachment-0001.html From brian at freeswitch.org Tue Jun 9 14:55:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 16:55:51 -0500 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <9615666.12391244584331944.JavaMail.daniel@radio> References: <9615666.12391244584331944.JavaMail.daniel@radio> Message-ID: <08122ADB-8711-418E-B580-033ACEC9339E@freeswitch.org> Someone hung the call up thats all I can imagine. Its not going to be us but maybe the far side you're talking to. Look at the sip trace bet the far end is sending you a BYE /b On Jun 9, 2009, at 4:52 PM, Dan wrote: > The freeswitch log show this before it hits my hangup hook in lua. > I don't have a pcap trace of the sip messaging, I can try that > tonight, it would definitely show where the hangup is coming from. > I'm recording to a local disk as a .ul file, so a headerless ulaw > format. > > 2009-06-08 21:25:44 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() > Channel sofia/external/5555551212 at 192.168.3.21 entering state > [terminated][200] > 2009-06-08 21:25:44 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() > Hangup sofia/external/5555551212 at 192.168.3.21 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-06-08 21:25:44 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal sofia/external/5555551212 at 192.168.3.21 > [KILL] > 2009-06-08 21:25:44 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal sofia/external/5555551212 at 192.168.3.21 > [BREAK] > 2009-06-08 21:25:44 [DEBUG] switch_core_codec.c:122 > switch_core_session_set_read_codec() Restore original codec. > 2009-06-08 21:25:44 [NOTICE] switch_cpp.cpp:1122 console_log() > hangupHook: 272423 status: hangup > > > > Thanks > D- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/80855e54/attachment.html From juanma.v82 at gmail.com Tue Jun 9 15:04:18 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Tue, 9 Jun 2009 19:04:18 -0300 Subject: [Freeswitch-users] Set Codec per Endpoint Message-ID: Hello, Is it posible to set a codec-pref per Endpoint instead to set it in sip-profiles? In my directory: I do this but FS do codec negotiation with all codecs in the profile internal. What i am doing wrong? Thx in advance From msc at freeswitch.org Tue Jun 9 15:06:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 15:06:01 -0700 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <08122ADB-8711-418E-B580-033ACEC9339E@freeswitch.org> References: <9615666.12391244584331944.JavaMail.daniel@radio> <08122ADB-8711-418E-B580-033ACEC9339E@freeswitch.org> Message-ID: <87f2f3b90906091506s5da8b925u256eaeaaf8c0fbb3@mail.gmail.com> On Tue, Jun 9, 2009 at 2:55 PM, Brian West wrote: > Someone hung the call up thats all I can imagine. Its not going to be us > but maybe the far side you're talking to. Look at the sip trace bet the far > end is sending you a BYE > This would make sense. Some systems will have a hard limit on the length of a call and will disconnect automatically because they interpret a 12 hour call as a "problem" and the "solution" is to hang up. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/87f29783/attachment.html From brian at freeswitch.org Tue Jun 9 15:10:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 17:10:07 -0500 Subject: [Freeswitch-users] Set Codec per Endpoint In-Reply-To: References: Message-ID: Do something like this Before you bridge /b On Jun 9, 2009, at 5:04 PM, JuanMa wrote: > codec-prefs Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/170d10ca/attachment.html From diego.viola at gmail.com Tue Jun 9 16:35:56 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 9 Jun 2009 19:35:56 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates Message-ID: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> Hi everyone, I'm trying to write a calling card app with A-Z rates, and I plan to use mod_lcr for this case... the only thing I need mod_lcr to do for me is get the rate based on the destination number / prefix. Is there a way I could achieve this with mod_lcr? I seen the wiki page and the SQL examples, but the SQL examples does a lot more, so I was thinking if I could use a custom SQL query to only do what I need. Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/e7a423e3/attachment.html From larclap at yahoo.com Tue Jun 9 16:43:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 16:43:03 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> Message-ID: <024701c9e95c$07e71ea0$17b55be0$@com> Thanks for the explanation, Brian; it was lost on me before. It was a DLink DIR-625 which had UPnP enabled. I turned it off. It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. 2009-06-09 16:24:32.271913 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-09 16:24:32.274131 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-09 16:24:32.663627 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-09 16:24:32.664053 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-09 16:24:32.913583 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-09 16:24:32.914581 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-09 16:24:33.415479 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-09 16:24:34.415249 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-09 16:24:36.413782 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-09 16:24:38.906588 [ERR] switch_nat.c:96 uPNP Device (url: http://192.168.10.253:4444/wipconn) returned an invalid external address of 0.0.0.0. Disabling uPNP 2009-06-09 16:24:38.906633 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! 2009-06-09 16:24:38.908650 [INFO] switch_core_sqldb.c:507 Opening DB 2009-06-09 16:24:38.950200 [NOTICE] switch_scheduler.c:166 Starting task thread Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 2:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls You have to start freeswitch without -nc to see it. Only happens during start up. /b On Jun 9, 2009, at 4:26 PM, Lars Zeb wrote: Rupa, Thanks for the detailed response. After upgrading from 13639 to 13732, I see no log errors. I am accessing Freeswitch vi fs_cli, but I did look in log/freeswitch.log. Certainly I see nothing that looks like your ERR below. I too have a dlink router. I will look at its configuration and see if upnp is enabled. Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8bd8a80e/attachment-0001.html From rupa at rupa.com Tue Jun 9 17:18:50 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 19:18:50 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <024701c9e95c$07e71ea0$17b55be0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> Message-ID: On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9d7fb251/attachment.html From larclap at yahoo.com Tue Jun 9 17:35:51 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 17:35:51 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> Message-ID: <027501c9e963$69e8fa40$3dbaeec0$@com> Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/7b625e35/attachment.html From rupa at rupa.com Tue Jun 9 17:37:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 19:37:24 -0500 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> Message-ID: Diego, Here is how I'd go about doing what I think you want. As payment, add a section to the wiki when you have it working. Create two profiles in lcr.conf. the first profile is your callingcard rate deck. Give it a profile id of 1. Then load your data into the lcr tables. carriers = define your carrier. call it whatever you want carrier_gateteway = you won't care about any real routes, so just load dummy data in here (linked to your carrier). lcr = load your rate deck here. Set profile id to 1. Now, to look up the customer's code, use the lcr application. application="lcr" data="$1 profilename" where profilename is the profile defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 format minus the + -- this is discussed in the wiki). At this point, you'll have the results of the lcr query in channel vars. ${lcr_route_count} tells you the number of routes found (if you loaded your rate deck right it should always be 1). ${lcr_rate_1} will contain the rate. So now you can set that to the var you need for nibblebill to work. If you want to use lcr to actually route the actual call, just call it again. This time with the profile id set to whatever you use to load the full lcr table for all your providers. On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: > Hi everyone, > > I'm trying to write a calling card app with A-Z rates, and I plan to use > mod_lcr for this case... the only thing I need mod_lcr to do for me is get > the rate based on the destination number / prefix. > > Is there a way I could achieve this with mod_lcr? I seen the wiki page and > the SQL examples, but the SQL examples does a lot more, so I was thinking if > I could use a custom SQL query to only do what I need. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9e6e5e47/attachment.html From diego.viola at gmail.com Tue Jun 9 17:52:38 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 9 Jun 2009 20:52:38 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> Message-ID: <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> Thanks for your help Rupa :). Don't worry that I will give everything back to the wiki, as I learn more and more, I have also contributed back some things to the wiki: http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola I love to do that, I will also contribute my calling card application to the community as soon as I'm done with it ;). Regards, Diego On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: > Diego, > > Here is how I'd go about doing what I think you want. As payment, add a > section to the wiki when you have it working. > > Create two profiles in lcr.conf. > > the first profile is your callingcard rate deck. Give it a profile id of > 1. Then load your data into the lcr tables. > > carriers = define your carrier. call it whatever you want > carrier_gateteway = you won't care about any real routes, so just load > dummy data in here (linked to your carrier). > lcr = load your rate deck here. Set profile id to 1. > > Now, to look up the customer's code, use the lcr application. > > application="lcr" data="$1 profilename" where profilename is the profile > defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 > format minus the + -- this is discussed in the wiki). > > At this point, you'll have the results of the lcr query in channel vars. > > ${lcr_route_count} tells you the number of routes found (if you loaded your > rate deck right it should always be 1). ${lcr_rate_1} will contain the > rate. > > So now you can set that to the var you need for nibblebill to work. > > If you want to use lcr to actually route the actual call, just call it > again. This time with the profile id set to whatever you use to load the > full lcr table for all your providers. > > On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: > >> Hi everyone, >> >> I'm trying to write a calling card app with A-Z rates, and I plan to use >> mod_lcr for this case... the only thing I need mod_lcr to do for me is get >> the rate based on the destination number / prefix. >> >> Is there a way I could achieve this with mod_lcr? I seen the wiki page and >> the SQL examples, but the SQL examples does a lot more, so I was thinking if >> I could use a custom SQL query to only do what I need. >> >> Thanks, >> >> Diego >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8ada7ec6/attachment.html From rupa at rupa.com Tue Jun 9 18:05:29 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 20:05:29 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <027501c9e963$69e8fa40$3dbaeec0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> Message-ID: if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: > Rupa, > > > > What options do I have for setting up logging? I?m sorry, but I don?t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 5:19 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > > > On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > > You might want to review how you have your logging setup then. The example > I gave you was copied/pasted out of my freeswitch.log file while testing > this fix. > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/3a070c79/attachment-0001.html From dujinfang at gmail.com Tue Jun 9 18:08:37 2009 From: dujinfang at gmail.com (seven) Date: Wed, 10 Jun 2009 09:08:37 +0800 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> Message-ID: <3DE17717-F722-488F-84C7-AFC805EE4137@gmail.com> On Jun 10, 2009, at 1:26 AM, freeswitch list wrote: > Awesome. Thank you so much guys. > > One last question. How would I forward a call in my dialplan to my > cell phone? > > I tried > > > > > but the caller id changes to the gateway callerid. > depend on your gateway provider, remember FreeSWITCH is a B2BUA. If your provider doesn't allow custom caller id, there's no way to ... > > On Tue, Jun 9, 2009 at 12:48 PM, dujinfang > wrote: >> 2) Is there anyway to have xml_curl send the password for directory >> entry requests. >> > > try this: > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/b71b9d44/attachment.html From larclap at yahoo.com Tue Jun 9 19:25:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 19:25:12 -0700 Subject: [Freeswitch-users] Documentation error in dialplan XML? Message-ID: <02a801c9e972$b07f0df0$117d29d0$@com> Is the closing of the condition element correct? I'm new at XML. Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/a9046edb/attachment.html From brian at freeswitch.org Tue Jun 9 19:32:04 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 21:32:04 -0500 Subject: [Freeswitch-users] Documentation error in dialplan XML? In-Reply-To: <02a801c9e972$b07f0df0$117d29d0$@com> References: <02a801c9e972$b07f0df0$117d29d0$@com> Message-ID: Nope shouldn't be there .. if you can update the wiki that would be great. /b On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote: > Is the closing of the condition element correct? I?m new at XML. > > > > > > should the slash at the end of the element be there? --> > > > > > > Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/5d8a2106/attachment.html From diego.viola at gmail.com Tue Jun 9 20:53:11 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 9 Jun 2009 23:53:11 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> Message-ID: <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> Hi everyone, I have used freeswitch/scripts/contrib/intralanman/C/lcr/sql/mysql-5.0.sql to load the mod_lcr schema, that worked well. But whenever I try to insert data from the "Sample Data" in the wiki it fails: http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data mysql> INSERT INTO lcr (digits, rate, carrier_id, lead_strip, trail_strip, -> prefix, suffix, -> date_start, date_end, quality, reliability) VALUES -> ('1', 0.15, 1, 0, 0, '', '', -> current_timestamp - interval 1 year, -> current_timestamp + interval 1 year -> , 0, 0); ERROR 1452 (23000): Cannot add or update a child row: a foreign key constraint fails (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) mysql> Regards, Diego On Tue, Jun 9, 2009 at 8:52 PM, Diego Viola wrote: > Thanks for your help Rupa :). > > Don't worry that I will give everything back to the wiki, as I learn more > and more, I have also contributed back some things to the wiki: > > http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola > > I love to do that, I will also contribute my calling card application to > the community as soon as I'm done with it ;). > > Regards, > > Diego > > > On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: > >> Diego, >> >> Here is how I'd go about doing what I think you want. As payment, add a >> section to the wiki when you have it working. >> >> Create two profiles in lcr.conf. >> >> the first profile is your callingcard rate deck. Give it a profile id of >> 1. Then load your data into the lcr tables. >> >> carriers = define your carrier. call it whatever you want >> carrier_gateteway = you won't care about any real routes, so just load >> dummy data in here (linked to your carrier). >> lcr = load your rate deck here. Set profile id to 1. >> >> Now, to look up the customer's code, use the lcr application. >> >> application="lcr" data="$1 profilename" where profilename is the profile >> defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 >> format minus the + -- this is discussed in the wiki). >> >> At this point, you'll have the results of the lcr query in channel vars. >> >> ${lcr_route_count} tells you the number of routes found (if you loaded >> your rate deck right it should always be 1). ${lcr_rate_1} will contain the >> rate. >> >> So now you can set that to the var you need for nibblebill to work. >> >> If you want to use lcr to actually route the actual call, just call it >> again. This time with the profile id set to whatever you use to load the >> full lcr table for all your providers. >> >> On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: >> >>> Hi everyone, >>> >>> I'm trying to write a calling card app with A-Z rates, and I plan to use >>> mod_lcr for this case... the only thing I need mod_lcr to do for me is get >>> the rate based on the destination number / prefix. >>> >>> Is there a way I could achieve this with mod_lcr? I seen the wiki page >>> and the SQL examples, but the SQL examples does a lot more, so I was >>> thinking if I could use a custom SQL query to only do what I need. >>> >>> Thanks, >>> >>> Diego >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -Rupa >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/212ad3b1/attachment-0001.html From mrene_lists at avgs.ca Tue Jun 9 20:55:00 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 9 Jun 2009 23:55:00 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> Message-ID: I think a foreign key constraint is failing, should look into that On 9-Jun-09, at 11:53 PM, Diego Viola wrote: > Hi everyone, > > I have used freeswitch/scripts/contrib/intralanman/C/lcr/sql/ > mysql-5.0.sql to load the mod_lcr schema, that worked well. > > But whenever I try to insert data from the "Sample Data" in the wiki > it fails: http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data > > mysql> INSERT INTO lcr (digits, rate, carrier_id, lead_strip, > trail_strip, > -> prefix, suffix, > -> date_start, date_end, quality, reliability) > VALUES > -> ('1', 0.15, 1, 0, 0, '', '', > -> current_timestamp - interval 1 year, > -> current_timestamp + interval 1 year > -> , 0, 0); > ERROR 1452 (23000): Cannot add or update a child row: a foreign key > constraint fails (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY > (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON > UPDATE CASCADE) > mysql> > > Regards, > > Diego > > On Tue, Jun 9, 2009 at 8:52 PM, Diego Viola > wrote: > Thanks for your help Rupa :). > > Don't worry that I will give everything back to the wiki, as I learn > more and more, I have also contributed back some things to the wiki: > > http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola > > I love to do that, I will also contribute my calling card > application to the community as soon as I'm done with it ;). > > Regards, > > Diego > > > On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: > Diego, > > Here is how I'd go about doing what I think you want. As payment, > add a section to the wiki when you have it working. > > Create two profiles in lcr.conf. > > the first profile is your callingcard rate deck. Give it a profile > id of 1. Then load your data into the lcr tables. > > carriers = define your carrier. call it whatever you want > carrier_gateteway = you won't care about any real routes, so just > load dummy data in here (linked to your carrier). > lcr = load your rate deck here. Set profile id to 1. > > Now, to look up the customer's code, use the lcr application. > > application="lcr" data="$1 profilename" where profilename is the > profile defined in lcr.conf with id 1. $1 is the normalized number > (I suggest e164 format minus the + -- this is discussed in the wiki). > > At this point, you'll have the results of the lcr query in channel > vars. > > ${lcr_route_count} tells you the number of routes found (if you > loaded your rate deck right it should always be 1). ${lcr_rate_1} > will contain the rate. > > So now you can set that to the var you need for nibblebill to work. > > If you want to use lcr to actually route the actual call, just call > it again. This time with the profile id set to whatever you use to > load the full lcr table for all your providers. > > On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola > wrote: > Hi everyone, > > I'm trying to write a calling card app with A-Z rates, and I plan to > use mod_lcr for this case... the only thing I need mod_lcr to do for > me is get the rate based on the destination number / prefix. > > Is there a way I could achieve this with mod_lcr? I seen the wiki > page and the SQL examples, but the SQL examples does a lot more, so > I was thinking if I could use a custom SQL query to only do what I > need. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/baa3225e/attachment.html From diego.viola at gmail.com Tue Jun 9 21:13:38 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 00:13:38 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> Message-ID: <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> Any other ideas? On Tue, Jun 9, 2009 at 11:55 PM, Mathieu Rene wrote: > I think a foreign key constraint is failing, should look into that > > On 9-Jun-09, at 11:53 PM, Diego Viola wrote: > > Hi everyone, > > I have used freeswitch/scripts/contrib/intralanman/C/lcr/sql/mysql-5.0.sql > to load the mod_lcr schema, that worked well. > > But whenever I try to insert data from the "Sample Data" in the wiki it > fails: http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data > > mysql> INSERT INTO lcr (digits, rate, carrier_id, lead_strip, trail_strip, > -> prefix, suffix, > -> date_start, date_end, quality, reliability) VALUES > -> ('1', 0.15, 1, 0, 0, '', '', > -> current_timestamp - interval 1 year, > -> current_timestamp + interval 1 year > -> , 0, 0); > ERROR 1452 (23000): Cannot add or update a child row: a foreign key > constraint fails (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY > (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE > CASCADE) > mysql> > > Regards, > > Diego > > On Tue, Jun 9, 2009 at 8:52 PM, Diego Viola wrote: > >> Thanks for your help Rupa :). >> >> Don't worry that I will give everything back to the wiki, as I learn more >> and more, I have also contributed back some things to the wiki: >> >> http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola >> >> I love to do that, I will also contribute my calling card application to >> the community as soon as I'm done with it ;). >> >> Regards, >> >> Diego >> >> >> On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: >> >>> Diego, >>> >>> Here is how I'd go about doing what I think you want. As payment, add a >>> section to the wiki when you have it working. >>> >>> Create two profiles in lcr.conf. >>> >>> the first profile is your callingcard rate deck. Give it a profile id of >>> 1. Then load your data into the lcr tables. >>> >>> carriers = define your carrier. call it whatever you want >>> carrier_gateteway = you won't care about any real routes, so just load >>> dummy data in here (linked to your carrier). >>> lcr = load your rate deck here. Set profile id to 1. >>> >>> Now, to look up the customer's code, use the lcr application. >>> >>> application="lcr" data="$1 profilename" where profilename is the profile >>> defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 >>> format minus the + -- this is discussed in the wiki). >>> >>> At this point, you'll have the results of the lcr query in channel vars. >>> >>> ${lcr_route_count} tells you the number of routes found (if you loaded >>> your rate deck right it should always be 1). ${lcr_rate_1} will contain the >>> rate. >>> >>> So now you can set that to the var you need for nibblebill to work. >>> >>> If you want to use lcr to actually route the actual call, just call it >>> again. This time with the profile id set to whatever you use to load the >>> full lcr table for all your providers. >>> >>> On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: >>> >>>> Hi everyone, >>>> >>>> I'm trying to write a calling card app with A-Z rates, and I plan to use >>>> mod_lcr for this case... the only thing I need mod_lcr to do for me is get >>>> the rate based on the destination number / prefix. >>>> >>>> Is there a way I could achieve this with mod_lcr? I seen the wiki page >>>> and the SQL examples, but the SQL examples does a lot more, so I was >>>> thinking if I could use a custom SQL query to only do what I need. >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -Rupa >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/93fd3452/attachment.html From larclap at yahoo.com Tue Jun 9 21:21:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 21:21:12 -0700 Subject: [Freeswitch-users] Documentation error in dialplan XML? In-Reply-To: References: <02a801c9e972$b07f0df0$117d29d0$@com> Message-ID: <02b901c9e982$e4ea2420$aebe6c60$@com> Done From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 7:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Documentation error in dialplan XML? Nope shouldn't be there .. if you can update the wiki that would be great. /b On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote: Is the closing of the condition element correct? I'm new at XML. Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/d8d3a3a1/attachment-0001.html From mrene_lists at avgs.ca Tue Jun 9 21:21:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 10 Jun 2009 00:21:48 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> Message-ID: If we look at the message again, (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) And put it into words: The contrainst names fs_lcr/lcr affecting field carrier_id, and referencing the id field of the carriers table, fails. In other words, the value you have for carrier_id does not match any value of id in the carriers table. Math On 10-Jun-09, at 12:13 AM, Diego Viola wrote: > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) From brian at freeswitch.org Tue Jun 9 21:23:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 23:23:15 -0500 Subject: [Freeswitch-users] Documentation error in dialplan XML? In-Reply-To: <02b901c9e982$e4ea2420$aebe6c60$@com> References: <02a801c9e972$b07f0df0$117d29d0$@com> <02b901c9e982$e4ea2420$aebe6c60$@com> Message-ID: <6428E0C1-1EF3-4219-8C51-18C8716D2105@freeswitch.org> Kewl, thanks! /b Sent from my iPhone On Jun 9, 2009, at 11:21 PM, "Lars Zeb" wrote: > Done > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: Tuesday, June 09, 2009 7:32 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Documentation error in dialplan XML? > > > > Nope shouldn't be there .. if you can update the wiki that would be > great. > > > > /b > > > > On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote: > > > > > Is the closing of the condition element correct? I?m new at XML. > > > > > > > > > > > > ?should the slash at the end of the element be there? --> > > > > > > > > > > > > Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/5a50a5c2/attachment.html From diego.viola at gmail.com Tue Jun 9 21:43:35 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 00:43:35 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> Message-ID: <86a32abc0906092143yb0262ddpd737c4fc45b9fa3c@mail.gmail.com> Fixed the issue, I will update the wiki now with a MySQL example. Regards, Diego On Wed, Jun 10, 2009 at 12:21 AM, Mathieu Rene wrote: > If we look at the message again, > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) > And put it into words: > The contrainst names fs_lcr/lcr affecting field carrier_id, and > referencing the id field of the carriers table, fails. > In other words, the value you have for carrier_id does not match any > value of id in the carriers table. > > Math > > On 10-Jun-09, at 12:13 AM, Diego Viola wrote: > > > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) > > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/10d23e68/attachment.html From diego.viola at gmail.com Tue Jun 9 22:07:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 01:07:47 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906092143yb0262ddpd737c4fc45b9fa3c@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> <86a32abc0906092143yb0262ddpd737c4fc45b9fa3c@mail.gmail.com> Message-ID: <86a32abc0906092207q3c669c44w38046c6289e41438@mail.gmail.com> Ok I have added a new MySQL example here. http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data Diego On Wed, Jun 10, 2009 at 12:43 AM, Diego Viola wrote: > Fixed the issue, I will update the wiki now with a MySQL example. > > Regards, > > Diego > > > On Wed, Jun 10, 2009 at 12:21 AM, Mathieu Rene wrote: > >> If we look at the message again, >> (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) >> REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) >> And put it into words: >> The contrainst names fs_lcr/lcr affecting field carrier_id, and >> referencing the id field of the carriers table, fails. >> In other words, the value you have for carrier_id does not match any >> value of id in the carriers table. >> >> Math >> >> On 10-Jun-09, at 12:13 AM, Diego Viola wrote: >> >> > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) >> > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/aa3dc0cf/attachment.html From shaheryarkh at googlemail.com Tue Jun 9 22:19:46 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 11:19:46 +0600 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Message-ID: Hi, What kind of problem you are referring to? I am using Skypiax from latest FS trunk revision no. 13613 on CentOS 5.3, Kernel 2.6.18-92.1.22.el5.centos.plusPAE without any problem, the system seems stable and going in production very soon. However, i would like to mention here that i have customized it a bit to add a couple of new commands to allow dynamic Skypiax interface addition and deletion in a running FreeSWITCH process, But instead of changing any existing code i have merely added new code to the exiting, so this shouldn't have resolved the problem you are referring to. The overall performance of both Skypiax and FS are excellent and we are extremely thankful to you guys for developing such great software. If you guys or anyone else need any help in setting up FS or Skypiax on CentOS, do write to me. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli wrote: > Hi all, > > there are problems for mod_skypiax in recent centos, with more than a > handful of concurrent Skype calls. > > Probably the problem is ALSA-related. > > Until it is solved, for production please use Ubuntu 8.04 (see below), > some other Linux distro (and please write here your experience), or > Windows. > > I modified the wiki page to reflect this ( > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > > If someone with CentOS knowledge can chime in I'll be grateful :-). > > Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > all infos, and feel free to contact me directly. > > -giovanni > > > > > > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/cf5ddde8/attachment-0001.html From shaheryarkh at googlemail.com Tue Jun 9 23:33:16 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 12:33:16 +0600 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Message-ID: Sorry, i didn't visited the Jira link you mentioned. Now i know the issue and I have replied it there. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli wrote: > Hi all, > > there are problems for mod_skypiax in recent centos, with more than a > handful of concurrent Skype calls. > > Probably the problem is ALSA-related. > > Until it is solved, for production please use Ubuntu 8.04 (see below), > some other Linux distro (and please write here your experience), or > Windows. > > I modified the wiki page to reflect this ( > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > > If someone with CentOS knowledge can chime in I'll be grateful :-). > > Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > all infos, and feel free to contact me directly. > > -giovanni > > > > > > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/f2512d0d/attachment.html From diego.viola at gmail.com Tue Jun 9 23:38:07 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 02:38:07 -0400 Subject: [Freeswitch-users] mod_nibblebill not set variable nibble_total_billed In-Reply-To: <200906091126.03556.yivzhenko@mksat.net> References: <200906091126.03556.yivzhenko@mksat.net> Message-ID: <86a32abc0906092338hc129269lb7417fc655380c27@mail.gmail.com> Open a Jira. http://jira.freeswitch.org/ On Tue, Jun 9, 2009 at 4:26 AM, Yuriy Ivzhenko wrote: > Some time ago mod_nibblebill was set variable nibble_total_billed after > hangup. > > But after last few updates of module this variable is no more sets. > > Somebody else have this problem? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/11a07e3c/attachment.html From gmaruzz at celliax.org Wed Jun 10 01:37:39 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 10:37:39 +0200 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Message-ID: <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> Ciao Muhammad! What a good news! Centos is the most stable and performing platform for FS, so I would really love to test and document on the wiki how to have a stable centos mod_skypiax installation. I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE ), and begin to test. In the mean time, do you have any hint, special procedure, etc you have done for having skypiax working well? Please, please, please let be in contact! :-) Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 8:33 AM, Muhammad Shahzad wrote: > Sorry, i didn't visited the Jira link you mentioned. Now i know the issue > and I have replied it there. > > Thank you. > > > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > wrote: >> >> Hi all, >> >> there are problems for mod_skypiax in recent centos, ?with more than a >> handful of concurrent Skype calls. >> >> Probably the problem is ALSA-related. >> >> Until it is solved, for production please use Ubuntu 8.04 (see below), >> some other Linux distro (and please write here your experience), or >> Windows. >> >> I modified the wiki page to reflect this ( >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for >> all infos, and feel free to contact me directly. >> >> -giovanni >> >> >> >> >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 02:16:25 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 15:16:25 +0600 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> Message-ID: Thanks. I didn't make any special arrangements for FS or Skypiax to work on CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE kernel with following commands, root ~# yum update root ~# yum install kernel-PAE i installed PAE kernel just because i wanted to increase System RAM to 8GB before i deploy it for production use, so i can double or even triple Skypiax channels whenever i need so, without system or FS shutdown. As far as a skypiax configuration is concerned, i did modified mod_skypiax.c to add a couple of commands to dynamically add and remove Skypiax interfaces in a running FS process. However, this code does not replaces or changes any previous code. Other then that there is no significant change in configuration steps. Though i did use mod_xml_curl to dynamically update skypiax interface configuration in FS. Thank you. On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad! > > What a good news! > > Centos is the most stable and performing platform for FS, so I would > really love to test and document on the wiki how to have a stable > centos mod_skypiax installation. > > I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE > ), and begin to test. In the mean time, do you have any hint, special > procedure, etc you have done for having skypiax working well? > > Please, please, please let be in contact! :-) > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > Shahzad wrote: > > Sorry, i didn't visited the Jira link you mentioned. Now i know the issue > > and I have replied it there. > > > > Thank you. > > > > > > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Hi all, > >> > >> there are problems for mod_skypiax in recent centos, with more than a > >> handful of concurrent Skype calls. > >> > >> Probably the problem is ALSA-related. > >> > >> Until it is solved, for production please use Ubuntu 8.04 (see below), > >> some other Linux distro (and please write here your experience), or > >> Windows. > >> > >> I modified the wiki page to reflect this ( > >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > >> > >> If someone with CentOS knowledge can chime in I'll be grateful :-). > >> > >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > >> all infos, and feel free to contact me directly. > >> > >> -giovanni > >> > >> > >> > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/a4c49161/attachment.html From gmaruzz at celliax.org Wed Jun 10 02:47:09 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 11:47:09 +0200 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> Message-ID: <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Ciao Muhammad, first thanks a lot for sharing your experience and help us in making a better software! >From the name of the kernel, seems that you are using centos5.2 is this correct? I just tried centos5.3 (64bit) with centosplus kernel, but no luck. I'm now installing a centos5.2 (64), I will test it with centosplus kernel and with its normal kernel. BTW, I would like *really* a lot to have and integrate your addition to the code (also if it needs some labor from me, no problem). Would you like to send it to me, so I will integrate in the main trunk and you don't have no more to maintain it? (so you can develop other cool features for mod_skypiax ;-) )? -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 11:16 AM, Muhammad Shahzad wrote: > Thanks. I didn't make any special arrangements for FS or Skypiax to work on > CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE > kernel with following commands, > > root ~# yum update > root ~# yum install kernel-PAE > > i installed PAE kernel just because i wanted to increase System RAM to 8GB > before i deploy it for production use, so i can double or even triple > Skypiax channels whenever i need so, without system or FS shutdown. > > As far as a skypiax configuration is concerned, i did modified mod_skypiax.c > to add a couple of commands to dynamically add and remove Skypiax interfaces > in a running FS process. However, this code does not replaces or changes any > previous code. Other then that there is no significant change in > configuration steps. Though i did use mod_xml_curl to dynamically update > skypiax interface configuration in FS. > > > Thank you. > > > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli > wrote: >> >> Ciao Muhammad! >> >> What a good news! >> >> Centos is the most stable and performing platform for FS, so I would >> really love to test and document on the wiki how to have a stable >> centos mod_skypiax installation. >> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >> ), and begin to test. In the mean time, do you have any hint, special >> procedure, etc you have done for having skypiax working well? >> >> Please, please, please let be in contact! :-) >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> Shahzad wrote: >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >> > issue >> > and I have replied it there. >> > >> > Thank you. >> > >> > >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> Hi all, >> >> >> >> there are problems for mod_skypiax in recent centos, ?with more than a >> >> handful of concurrent Skype calls. >> >> >> >> Probably the problem is ALSA-related. >> >> >> >> Until it is solved, for production please use Ubuntu 8.04 (see below), >> >> some other Linux distro (and please write here your experience), or >> >> Windows. >> >> >> >> I modified the wiki page to reflect this ( >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for >> >> all infos, and feel free to contact me directly. >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 03:47:38 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 16:47:38 +0600 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable kernel. I have heard 64bit ALSA drivers have bad sound issues, but never used it personally. As for source code of my modifications, i made those change to develop a customized commercial solution for large European firm, so i would need their permissions to provide you the required official patch. Let me write them an offical request for this. Thank you. On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad, > > first thanks a lot for sharing your experience and help us in making a > better software! > > >From the name of the kernel, seems that you are using centos5.2 is this > correct? > > I just tried centos5.3 (64bit) with centosplus kernel, but no luck. > > I'm now installing a centos5.2 (64), I will test it with centosplus > kernel and with its normal kernel. > > BTW, I would like *really* a lot to have and integrate your addition > to the code (also if it needs some labor from me, no problem). Would > you like to send it to me, so I will integrate in the main trunk and > you don't have no more to maintain it? (so you can develop other cool > features for mod_skypiax ;-) )? > > -giovanni > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 11:16 AM, Muhammad > Shahzad wrote: > > Thanks. I didn't make any special arrangements for FS or Skypiax to work > on > > CentOS 5.3. I only enabled CentOS Plus yum repository and then install > PAE > > kernel with following commands, > > > > root ~# yum update > > root ~# yum install kernel-PAE > > > > i installed PAE kernel just because i wanted to increase System RAM to > 8GB > > before i deploy it for production use, so i can double or even triple > > Skypiax channels whenever i need so, without system or FS shutdown. > > > > As far as a skypiax configuration is concerned, i did modified > mod_skypiax.c > > to add a couple of commands to dynamically add and remove Skypiax > interfaces > > in a running FS process. However, this code does not replaces or changes > any > > previous code. Other then that there is no significant change in > > configuration steps. Though i did use mod_xml_curl to dynamically update > > skypiax interface configuration in FS. > > > > > > Thank you. > > > > > > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Ciao Muhammad! > >> > >> What a good news! > >> > >> Centos is the most stable and performing platform for FS, so I would > >> really love to test and document on the wiki how to have a stable > >> centos mod_skypiax installation. > >> > >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE > >> ), and begin to test. In the mean time, do you have any hint, special > >> procedure, etc you have done for having skypiax working well? > >> > >> Please, please, please let be in contact! :-) > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> > >> > >> > >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > >> Shahzad wrote: > >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the > >> > issue > >> > and I have replied it there. > >> > > >> > Thank you. > >> > > >> > > >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > >> > > >> > wrote: > >> >> > >> >> Hi all, > >> >> > >> >> there are problems for mod_skypiax in recent centos, with more than > a > >> >> handful of concurrent Skype calls. > >> >> > >> >> Probably the problem is ALSA-related. > >> >> > >> >> Until it is solved, for production please use Ubuntu 8.04 (see > below), > >> >> some other Linux distro (and please write here your experience), or > >> >> Windows. > >> >> > >> >> I modified the wiki page to reflect this ( > >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > >> >> > >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). > >> >> > >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > >> >> all infos, and feel free to contact me directly. > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> ========================================= > >> >> www.celliax.org > >> >> via Pierlombardo 9, 20135 Milano > >> >> Italy > >> >> gmaruzz at celliax dot org > >> >> Cell : +39-347-2665618 > >> >> Fax : +39-02-87390039 > >> >> > >> >> _______________________________________________ > >> >> Freeswitch-users mailing list > >> >> Freeswitch-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Muhammad Shahzad > >> > ----------------------------------- > >> > CISCO Rich Media Communication Specialist (CRMCS) > >> > CISCO Certified Network Associate (CCNA) > >> > Cell: +92 334 422 40 88 > >> > MSN: shari_786pk at hotmail.com > >> > Email: shaheryarkh at googlemail.com > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/e507a830/attachment.html From gmaruzz at celliax.org Wed Jun 10 05:27:46 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 14:27:46 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: <7b197bef0906100527w4876413bif292a7dcadf60fd4@mail.gmail.com> Thanks a lot Muhammad, and please let that firm know the advantages of having customizations included into mainstream ;-). OK, I will try 32 bit too, and see if there is differences. So, you started fom a fresh install of centos5.2, then you installed the PAE kernel. Is this right? Pay attention, because if you do an "yum update" now, it will install the "128" kernel, no more the "92", and maybe this will break something..... Anyway, I'm investigating, and please let me know if you'll have additional infos. Hope to hear from you soon, -giovanni On Wed, Jun 10, 2009 at 12:47 PM, Muhammad Shahzad wrote: > I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable > kernel. I have heard 64bit ALSA drivers have bad sound issues, but never > used it personally. > > As for source code of my modifications, i made those change to develop a > customized commercial solution for large European firm, so i would need > their permissions to provide you the required official patch. Let me write > them an offical request for this. > > Thank you. > > > On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli > wrote: >> >> Ciao Muhammad, >> >> first thanks a lot for sharing your experience and help us in making a >> better software! >> >> >From the name of the kernel, seems that you are using centos5.2 is this >> correct? >> >> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >> >> I'm now installing a centos5.2 (64), I will test it with centosplus >> kernel and with its normal kernel. >> >> BTW, I would like *really* a lot to have and integrate your addition >> to the code (also if it needs some labor from me, no problem). Would >> you like to send it to me, so I will integrate in the main trunk and >> you don't have no more to maintain it? (so you can develop other cool >> features for mod_skypiax ;-) )? >> >> -giovanni >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >> Shahzad wrote: >> > Thanks. I didn't make any special arrangements for FS or Skypiax to work >> > on >> > CentOS 5.3. I only enabled CentOS Plus yum repository and then install >> > PAE >> > kernel with following commands, >> > >> > root ~# yum update >> > root ~# yum install kernel-PAE >> > >> > i installed PAE kernel just because i wanted to increase System RAM to >> > 8GB >> > before i deploy it for production use, so i can double or even triple >> > Skypiax channels whenever i need so, without system or FS shutdown. >> > >> > As far as a skypiax configuration is concerned, i did modified >> > mod_skypiax.c >> > to add a couple of commands to dynamically add and remove Skypiax >> > interfaces >> > in a running FS process. However, this code does not replaces or changes >> > any >> > previous code. Other then that there is no significant change in >> > configuration steps. Though i did use mod_xml_curl to dynamically update >> > skypiax interface configuration in FS. >> > >> > >> > Thank you. >> > >> > >> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> Ciao Muhammad! >> >> >> >> What a good news! >> >> >> >> Centos is the most stable and performing platform for FS, so I would >> >> really love to test and document on the wiki how to have a stable >> >> centos mod_skypiax installation. >> >> >> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >> >> ), and begin to test. In the mean time, do you have any hint, special >> >> procedure, etc you have done for having skypiax working well? >> >> >> >> Please, please, please let be in contact! :-) >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> >> >> >> >> >> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> >> Shahzad wrote: >> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >> >> > issue >> >> > and I have replied it there. >> >> > >> >> > Thank you. >> >> > >> >> > >> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> >> > >> >> > wrote: >> >> >> >> >> >> Hi all, >> >> >> >> >> >> there are problems for mod_skypiax in recent centos, ?with more than >> >> >> a >> >> >> handful of concurrent Skype calls. >> >> >> >> >> >> Probably the problem is ALSA-related. >> >> >> >> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >> >> >> below), >> >> >> some other Linux distro (and please write here your experience), or >> >> >> Windows. >> >> >> >> >> >> I modified the wiki page to reflect this ( >> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> >> >> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> >> >> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for >> >> >> all infos, and feel free to contact me directly. >> >> >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> >> >> Giovanni Maruzzelli >> >> >> ========================================= >> >> >> www.celliax.org >> >> >> via Pierlombardo 9, 20135 Milano >> >> >> Italy >> >> >> gmaruzz at celliax dot org >> >> >> Cell : +39-347-2665618 >> >> >> Fax : +39-02-87390039 >> >> >> >> >> >> _______________________________________________ >> >> >> Freeswitch-users mailing list >> >> >> Freeswitch-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Muhammad Shahzad >> >> > ----------------------------------- >> >> > CISCO Rich Media Communication Specialist (CRMCS) >> >> > CISCO Certified Network Associate (CCNA) >> >> > Cell: +92 334 422 40 88 >> >> > MSN: shari_786pk at hotmail.com >> >> > Email: shaheryarkh at googlemail.com >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 05:29:44 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 18:29:44 +0600 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: I am glad to share the patch to enable dynamic Skypiax interfaces in FS. Please do note that however, that i started working on it on May 22, 2009. So any officaily changes made to mod_skypiax.c since then will not appear in it and will be lost if you apply this patch blindly. I request Giovanni Maruzzelli to carefully merge this patch in main stream code before committing it to FS SVN. Thank you. On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable > kernel. I have heard 64bit ALSA drivers have bad sound issues, but never > used it personally. > > As for source code of my modifications, i made those change to develop a > customized commercial solution for large European firm, so i would need > their permissions to provide you the required official patch. Let me write > them an offical request for this. > > Thank you. > > > > On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli wrote: > >> Ciao Muhammad, >> >> first thanks a lot for sharing your experience and help us in making a >> better software! >> >> >From the name of the kernel, seems that you are using centos5.2 is this >> correct? >> >> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >> >> I'm now installing a centos5.2 (64), I will test it with centosplus >> kernel and with its normal kernel. >> >> BTW, I would like *really* a lot to have and integrate your addition >> to the code (also if it needs some labor from me, no problem). Would >> you like to send it to me, so I will integrate in the main trunk and >> you don't have no more to maintain it? (so you can develop other cool >> features for mod_skypiax ;-) )? >> >> -giovanni >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >> Shahzad wrote: >> > Thanks. I didn't make any special arrangements for FS or Skypiax to work >> on >> > CentOS 5.3. I only enabled CentOS Plus yum repository and then install >> PAE >> > kernel with following commands, >> > >> > root ~# yum update >> > root ~# yum install kernel-PAE >> > >> > i installed PAE kernel just because i wanted to increase System RAM to >> 8GB >> > before i deploy it for production use, so i can double or even triple >> > Skypiax channels whenever i need so, without system or FS shutdown. >> > >> > As far as a skypiax configuration is concerned, i did modified >> mod_skypiax.c >> > to add a couple of commands to dynamically add and remove Skypiax >> interfaces >> > in a running FS process. However, this code does not replaces or changes >> any >> > previous code. Other then that there is no significant change in >> > configuration steps. Though i did use mod_xml_curl to dynamically update >> > skypiax interface configuration in FS. >> > >> > >> > Thank you. >> > >> > >> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli < >> gmaruzz at celliax.org> >> > wrote: >> >> >> >> Ciao Muhammad! >> >> >> >> What a good news! >> >> >> >> Centos is the most stable and performing platform for FS, so I would >> >> really love to test and document on the wiki how to have a stable >> >> centos mod_skypiax installation. >> >> >> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >> >> ), and begin to test. In the mean time, do you have any hint, special >> >> procedure, etc you have done for having skypiax working well? >> >> >> >> Please, please, please let be in contact! :-) >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> >> >> >> >> >> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> >> Shahzad wrote: >> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >> >> > issue >> >> > and I have replied it there. >> >> > >> >> > Thank you. >> >> > >> >> > >> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> >> > >> >> > wrote: >> >> >> >> >> >> Hi all, >> >> >> >> >> >> there are problems for mod_skypiax in recent centos, with more than >> a >> >> >> handful of concurrent Skype calls. >> >> >> >> >> >> Probably the problem is ALSA-related. >> >> >> >> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >> below), >> >> >> some other Linux distro (and please write here your experience), or >> >> >> Windows. >> >> >> >> >> >> I modified the wiki page to reflect this ( >> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> >> >> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> >> >> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34for >> >> >> all infos, and feel free to contact me directly. >> >> >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> >> >> Giovanni Maruzzelli >> >> >> ========================================= >> >> >> www.celliax.org >> >> >> via Pierlombardo 9, 20135 Milano >> >> >> Italy >> >> >> gmaruzz at celliax dot org >> >> >> Cell : +39-347-2665618 >> >> >> Fax : +39-02-87390039 >> >> >> >> >> >> _______________________________________________ >> >> >> Freeswitch-users mailing list >> >> >> Freeswitch-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Muhammad Shahzad >> >> > ----------------------------------- >> >> > CISCO Rich Media Communication Specialist (CRMCS) >> >> > CISCO Certified Network Associate (CCNA) >> >> > Cell: +92 334 422 40 88 >> >> > MSN: shari_786pk at hotmail.com >> >> > Email: shaheryarkh at googlemail.com >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/efce2963/attachment.html From gmaruzz at celliax.org Wed Jun 10 05:33:34 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 14:33:34 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> Ciao Muhammad, you're faster than light :-)! the patch will be integrated very soon, I'll let you know when I'm done it. Keep enhancements, patches, bug fixes, etc flowing! thanks again, and thanks to the firm that so quickly understood and authorized you, -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 2:29 PM, Muhammad Shahzad wrote: > I am glad to share the patch to enable dynamic Skypiax interfaces in FS. > Please do note that however, that i started working on it on May 22, 2009. > So any officaily changes made to mod_skypiax.c since then will not appear in > it and will be lost if you apply this patch blindly. > > I request Giovanni Maruzzelli to carefully merge this patch in main stream > code before committing it to FS SVN. > > Thank you. > > > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad > wrote: >> >> I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable >> kernel. I have heard 64bit ALSA drivers have bad sound issues, but never >> used it personally. >> >> As for source code of my modifications, i made those change to develop a >> customized commercial solution for large European firm, so i would need >> their permissions to provide you the required official patch. Let me write >> them an offical request for this. >> >> Thank you. >> >> >> On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli >> wrote: >>> >>> Ciao Muhammad, >>> >>> first thanks a lot for sharing your experience and help us in making a >>> better software! >>> >>> >From the name of the kernel, seems that you are using centos5.2 is this >>> correct? >>> >>> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >>> >>> I'm now installing a centos5.2 (64), I will test it with centosplus >>> kernel and with its normal kernel. >>> >>> BTW, I would like *really* a lot to have and integrate your addition >>> to the code (also if it needs some labor from me, no problem). Would >>> you like to send it to me, so I will integrate in the main trunk and >>> you don't have no more to maintain it? (so you can develop other cool >>> features for mod_skypiax ;-) )? >>> >>> -giovanni >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> ========================================= >>> www.celliax.org >>> via Pierlombardo 9, 20135 Milano >>> Italy >>> gmaruzz at celliax dot org >>> Cell : +39-347-2665618 >>> Fax : +39-02-87390039 >>> >>> >>> >>> >>> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >>> Shahzad wrote: >>> > Thanks. I didn't make any special arrangements for FS or Skypiax to >>> > work on >>> > CentOS 5.3. I only enabled CentOS Plus yum repository and then install >>> > PAE >>> > kernel with following commands, >>> > >>> > root ~# yum update >>> > root ~# yum install kernel-PAE >>> > >>> > i installed PAE kernel just because i wanted to increase System RAM to >>> > 8GB >>> > before i deploy it for production use, so i can double or even triple >>> > Skypiax channels whenever i need so, without system or FS shutdown. >>> > >>> > As far as a skypiax configuration is concerned, i did modified >>> > mod_skypiax.c >>> > to add a couple of commands to dynamically add and remove Skypiax >>> > interfaces >>> > in a running FS process. However, this code does not replaces or >>> > changes any >>> > previous code. Other then that there is no significant change in >>> > configuration steps. Though i did use mod_xml_curl to dynamically >>> > update >>> > skypiax interface configuration in FS. >>> > >>> > >>> > Thank you. >>> > >>> > >>> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli >>> > >>> > wrote: >>> >> >>> >> Ciao Muhammad! >>> >> >>> >> What a good news! >>> >> >>> >> Centos is the most stable and performing platform for FS, so I would >>> >> really love to test and document on the wiki how to have a stable >>> >> centos mod_skypiax installation. >>> >> >>> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >>> >> ), and begin to test. In the mean time, do you have any hint, special >>> >> procedure, etc you have done for having skypiax working well? >>> >> >>> >> Please, please, please let be in contact! :-) >>> >> >>> >> >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> ========================================= >>> >> www.celliax.org >>> >> via Pierlombardo 9, 20135 Milano >>> >> Italy >>> >> gmaruzz at celliax dot org >>> >> Cell : +39-347-2665618 >>> >> Fax : +39-02-87390039 >>> >> >>> >> >>> >> >>> >> >>> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >>> >> Shahzad wrote: >>> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >>> >> > issue >>> >> > and I have replied it there. >>> >> > >>> >> > Thank you. >>> >> > >>> >> > >>> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >>> >> > >>> >> > wrote: >>> >> >> >>> >> >> Hi all, >>> >> >> >>> >> >> there are problems for mod_skypiax in recent centos, ?with more >>> >> >> than a >>> >> >> handful of concurrent Skype calls. >>> >> >> >>> >> >> Probably the problem is ALSA-related. >>> >> >> >>> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >>> >> >> below), >>> >> >> some other Linux distro (and please write here your experience), or >>> >> >> Windows. >>> >> >> >>> >> >> I modified the wiki page to reflect this ( >>> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >>> >> >> >>> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >>> >> >> >>> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 >>> >> >> for >>> >> >> all infos, and feel free to contact me directly. >>> >> >> >>> >> >> -giovanni >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> Sincerely, >>> >> >> >>> >> >> Giovanni Maruzzelli >>> >> >> ========================================= >>> >> >> www.celliax.org >>> >> >> via Pierlombardo 9, 20135 Milano >>> >> >> Italy >>> >> >> gmaruzz at celliax dot org >>> >> >> Cell : +39-347-2665618 >>> >> >> Fax : +39-02-87390039 >>> >> >> >>> >> >> _______________________________________________ >>> >> >> Freeswitch-users mailing list >>> >> >> Freeswitch-users at lists.freeswitch.org >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> >>> >> >> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> http://www.freeswitch.org >>> >> > >>> >> > >>> >> > >>> >> > -- >>> >> > Muhammad Shahzad >>> >> > ----------------------------------- >>> >> > CISCO Rich Media Communication Specialist (CRMCS) >>> >> > CISCO Certified Network Associate (CCNA) >>> >> > Cell: +92 334 422 40 88 >>> >> > MSN: shari_786pk at hotmail.com >>> >> > Email: shaheryarkh at googlemail.com >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > -- >>> > Muhammad Shahzad >>> > ----------------------------------- >>> > CISCO Rich Media Communication Specialist (CRMCS) >>> > CISCO Certified Network Associate (CCNA) >>> > Cell: +92 334 422 40 88 >>> > MSN: shari_786pk at hotmail.com >>> > Email: shaheryarkh at googlemail.com >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> _______________________________________________ >>> Freeswitch-dev mailing list >>> Freeswitch-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 06:01:57 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 19:01:57 +0600 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> Message-ID: You are welcome. Let me elaborate my setup here, I have two machines, one for development, this is basically my lenovo 3000 N200 laptop, it has following specs, 1. Intel 1.6 GHz with 1GB RAM. 2. CentOS 5.3 with Kernel 2.6.18-128.1.6.el5. 3. FS SVN revision Revision ID 13613. root ~# uname -a Linux localhost.localdomain 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:19:18 EDT 2009 i686 i686 i386 GNU/Linux root ~# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m I am using this machine extensively for my development projects, including Skypiax. Yesterday i gave a presentation to the board of directors of the said firm, regarding existing status of my project. They tested the setup with 2-3 concurrent SIP -> SKYPIAX and then SKYPIAX to SIP calls without any problem. So, i believe this configuration works without any sound issue...! The second machine is my test machine in a remote data center. I didn't prepare this machine, however, from SSH console i can see it has following specs, 1. Intel(R) Xeon(R) CPU E5405 @ 2.00GHz with 4 GB of RAM. 2. CentOS 5.3 with kernel 2.6.18-92.1.22.el5.centos.plusPAE. 3. FS SVN revision Revision ID 13613. root ~# uname -a Linux localhost.localdomain 2.6.18-92.1.22.el5.centos.plusPAE #1 SMP Wed Dec 17 11:32:56 EST 2008 i686 i686 i386 GNU/Linux root ~# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m Each machine that i use always, get update with yum update command BEFORE i do anything else on it. Hope this info will be helpful for you. Can you give me step by step procedure of your testing that is producing this bad sound result? I would like to perform this test on my both machines and see if i get the same results too. Thank you. On Wed, Jun 10, 2009 at 6:33 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad, > you're faster than light :-)! > > the patch will be integrated very soon, I'll let you know when I'm done it. > > Keep enhancements, patches, bug fixes, etc flowing! > > thanks again, and thanks to the firm that so quickly understood and > authorized you, > > -giovanni > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 2:29 PM, Muhammad > Shahzad wrote: > > I am glad to share the patch to enable dynamic Skypiax interfaces in FS. > > Please do note that however, that i started working on it on May 22, > 2009. > > So any officaily changes made to mod_skypiax.c since then will not appear > in > > it and will be lost if you apply this patch blindly. > > > > I request Giovanni Maruzzelli to carefully merge this patch in main > stream > > code before committing it to FS SVN. > > > > Thank you. > > > > > > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad > > wrote: > >> > >> I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable > >> kernel. I have heard 64bit ALSA drivers have bad sound issues, but never > >> used it personally. > >> > >> As for source code of my modifications, i made those change to develop a > >> customized commercial solution for large European firm, so i would need > >> their permissions to provide you the required official patch. Let me > write > >> them an offical request for this. > >> > >> Thank you. > >> > >> > >> On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > >> wrote: > >>> > >>> Ciao Muhammad, > >>> > >>> first thanks a lot for sharing your experience and help us in making a > >>> better software! > >>> > >>> >From the name of the kernel, seems that you are using centos5.2 is > this > >>> correct? > >>> > >>> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. > >>> > >>> I'm now installing a centos5.2 (64), I will test it with centosplus > >>> kernel and with its normal kernel. > >>> > >>> BTW, I would like *really* a lot to have and integrate your addition > >>> to the code (also if it needs some labor from me, no problem). Would > >>> you like to send it to me, so I will integrate in the main trunk and > >>> you don't have no more to maintain it? (so you can develop other cool > >>> features for mod_skypiax ;-) )? > >>> > >>> -giovanni > >>> > >>> Sincerely, > >>> > >>> Giovanni Maruzzelli > >>> ========================================= > >>> www.celliax.org > >>> via Pierlombardo 9, 20135 Milano > >>> Italy > >>> gmaruzz at celliax dot org > >>> Cell : +39-347-2665618 > >>> Fax : +39-02-87390039 > >>> > >>> > >>> > >>> > >>> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad > >>> Shahzad wrote: > >>> > Thanks. I didn't make any special arrangements for FS or Skypiax to > >>> > work on > >>> > CentOS 5.3. I only enabled CentOS Plus yum repository and then > install > >>> > PAE > >>> > kernel with following commands, > >>> > > >>> > root ~# yum update > >>> > root ~# yum install kernel-PAE > >>> > > >>> > i installed PAE kernel just because i wanted to increase System RAM > to > >>> > 8GB > >>> > before i deploy it for production use, so i can double or even triple > >>> > Skypiax channels whenever i need so, without system or FS shutdown. > >>> > > >>> > As far as a skypiax configuration is concerned, i did modified > >>> > mod_skypiax.c > >>> > to add a couple of commands to dynamically add and remove Skypiax > >>> > interfaces > >>> > in a running FS process. However, this code does not replaces or > >>> > changes any > >>> > previous code. Other then that there is no significant change in > >>> > configuration steps. Though i did use mod_xml_curl to dynamically > >>> > update > >>> > skypiax interface configuration in FS. > >>> > > >>> > > >>> > Thank you. > >>> > > >>> > > >>> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli > >>> > > >>> > wrote: > >>> >> > >>> >> Ciao Muhammad! > >>> >> > >>> >> What a good news! > >>> >> > >>> >> Centos is the most stable and performing platform for FS, so I would > >>> >> really love to test and document on the wiki how to have a stable > >>> >> centos mod_skypiax installation. > >>> >> > >>> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE > >>> >> ), and begin to test. In the mean time, do you have any hint, > special > >>> >> procedure, etc you have done for having skypiax working well? > >>> >> > >>> >> Please, please, please let be in contact! :-) > >>> >> > >>> >> > >>> >> Sincerely, > >>> >> > >>> >> Giovanni Maruzzelli > >>> >> ========================================= > >>> >> www.celliax.org > >>> >> via Pierlombardo 9, 20135 Milano > >>> >> Italy > >>> >> gmaruzz at celliax dot org > >>> >> Cell : +39-347-2665618 > >>> >> Fax : +39-02-87390039 > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > >>> >> Shahzad wrote: > >>> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know > the > >>> >> > issue > >>> >> > and I have replied it there. > >>> >> > > >>> >> > Thank you. > >>> >> > > >>> >> > > >>> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > >>> >> > > >>> >> > wrote: > >>> >> >> > >>> >> >> Hi all, > >>> >> >> > >>> >> >> there are problems for mod_skypiax in recent centos, with more > >>> >> >> than a > >>> >> >> handful of concurrent Skype calls. > >>> >> >> > >>> >> >> Probably the problem is ALSA-related. > >>> >> >> > >>> >> >> Until it is solved, for production please use Ubuntu 8.04 (see > >>> >> >> below), > >>> >> >> some other Linux distro (and please write here your experience), > or > >>> >> >> Windows. > >>> >> >> > >>> >> >> I modified the wiki page to reflect this ( > >>> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk) > >>> >> >> > >>> >> >> If someone with CentOS knowledge can chime in I'll be grateful > :-). > >>> >> >> > >>> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 > >>> >> >> for > >>> >> >> all infos, and feel free to contact me directly. > >>> >> >> > >>> >> >> -giovanni > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> Sincerely, > >>> >> >> > >>> >> >> Giovanni Maruzzelli > >>> >> >> ========================================= > >>> >> >> www.celliax.org > >>> >> >> via Pierlombardo 9, 20135 Milano > >>> >> >> Italy > >>> >> >> gmaruzz at celliax dot org > >>> >> >> Cell : +39-347-2665618 > >>> >> >> Fax : +39-02-87390039 > >>> >> >> > >>> >> >> _______________________________________________ > >>> >> >> Freeswitch-users mailing list > >>> >> >> Freeswitch-users at lists.freeswitch.org > >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> >> > >>> >> >> > >>> >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> >> http://www.freeswitch.org > >>> >> > > >>> >> > > >>> >> > > >>> >> > -- > >>> >> > Muhammad Shahzad > >>> >> > ----------------------------------- > >>> >> > CISCO Rich Media Communication Specialist (CRMCS) > >>> >> > CISCO Certified Network Associate (CCNA) > >>> >> > Cell: +92 334 422 40 88 > >>> >> > MSN: shari_786pk at hotmail.com > >>> >> > Email: shaheryarkh at googlemail.com > >>> >> > > >>> >> > _______________________________________________ > >>> >> > Freeswitch-users mailing list > >>> >> > Freeswitch-users at lists.freeswitch.org > >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > > >>> >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> > http://www.freeswitch.org > >>> >> > > >>> >> > > >>> >> > >>> >> _______________________________________________ > >>> >> Freeswitch-users mailing list > >>> >> Freeswitch-users at lists.freeswitch.org > >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> http://www.freeswitch.org > >>> > > >>> > > >>> > > >>> > -- > >>> > Muhammad Shahzad > >>> > ----------------------------------- > >>> > CISCO Rich Media Communication Specialist (CRMCS) > >>> > CISCO Certified Network Associate (CCNA) > >>> > Cell: +92 334 422 40 88 > >>> > MSN: shari_786pk at hotmail.com > >>> > Email: shaheryarkh at googlemail.com > >>> > > >>> > _______________________________________________ > >>> > Freeswitch-users mailing list > >>> > Freeswitch-users at lists.freeswitch.org > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > > >>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > > >>> > > >>> > >>> _______________________________________________ > >>> Freeswitch-dev mailing list > >>> Freeswitch-dev at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > >>> http://www.freeswitch.org > >> > >> > >> > >> -- > >> Muhammad Shahzad > >> ----------------------------------- > >> CISCO Rich Media Communication Specialist (CRMCS) > >> CISCO Certified Network Associate (CCNA) > >> Cell: +92 334 422 40 88 > >> MSN: shari_786pk at hotmail.com > >> Email: shaheryarkh at googlemail.com > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/8ff45edc/attachment-0001.html From teqx at yahoo.com Wed Jun 10 06:05:03 2009 From: teqx at yahoo.com (zigurds) Date: Wed, 10 Jun 2009 06:05:03 -0700 (PDT) Subject: [Freeswitch-users] how to determine who is hangup the call Message-ID: <23961888.post@talk.nabble.com> Hi, How to determine from channel events, that are sent through event_socket, what party have terminated the call? If I call for example from 103 to 104 and in first time 104 hang up, but in second time 103 hang up, in both cases CHANNEL_HANGUP is sent at first to 104: Event-Name: CHANNEL_HANGUP Core-UUID: 8729f181-3325-4373-8030-537bc090e9d3 ... Event-Date-Timestamp: 1244634602734687 Event-Calling-File: switch_core_state_machine.c Event-Calling-Function: switch_core_session_run Event-Calling-Line-Number: 469 Hangup-Cause: NORMAL_CLEARING Channel-State: CS_HANGUP Channel-State-Number: 10 Channel-Name: sofia/Test/104 Unique-ID: 17cdd5b7-8949-44d4-bbb9-5d17a713811c Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered ... Thanks, Zigurds -- View this message in context: http://www.nabble.com/how-to-determine-who-is-hangup-the-call-tp23961888p23961888.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Wed Jun 10 06:10:50 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 21:10:50 +0800 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: Glad to see the patch, I have been waiting for a long time :) btw, I have changed the start scripts a bit to start X and skype separately, glad to share it if someone interested. On Jun 10, 2009, at 8:29 PM, Muhammad Shahzad wrote: > I am glad to share the patch to enable dynamic Skypiax interfaces in > FS. Please do note that however, that i started working on it on May > 22, 2009. So any officaily changes made to mod_skypiax.c since then > will not appear in it and will be lost if you apply this patch > blindly. > > I request Giovanni Maruzzelli to carefully merge this patch in main > stream code before committing it to FS SVN. > > Thank you. > > > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad > wrote: > I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE > enable kernel. I have heard 64bit ALSA drivers have bad sound > issues, but never used it personally. > > As for source code of my modifications, i made those change to > develop a customized commercial solution for large European firm, so > i would need their permissions to provide you the required official > patch. Let me write them an offical request for this. > > Thank you. > > > > On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli > wrote: > Ciao Muhammad, > > first thanks a lot for sharing your experience and help us in making a > better software! > > >From the name of the kernel, seems that you are using centos5.2 is > this correct? > > I just tried centos5.3 (64bit) with centosplus kernel, but no luck. > > I'm now installing a centos5.2 (64), I will test it with centosplus > kernel and with its normal kernel. > > BTW, I would like *really* a lot to have and integrate your addition > to the code (also if it needs some labor from me, no problem). Would > you like to send it to me, so I will integrate in the main trunk and > you don't have no more to maintain it? (so you can develop other cool > features for mod_skypiax ;-) )? > > -giovanni > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 11:16 AM, Muhammad > Shahzad wrote: > > Thanks. I didn't make any special arrangements for FS or Skypiax > to work on > > CentOS 5.3. I only enabled CentOS Plus yum repository and then > install PAE > > kernel with following commands, > > > > root ~# yum update > > root ~# yum install kernel-PAE > > > > i installed PAE kernel just because i wanted to increase System > RAM to 8GB > > before i deploy it for production use, so i can double or even > triple > > Skypiax channels whenever i need so, without system or FS shutdown. > > > > As far as a skypiax configuration is concerned, i did modified > mod_skypiax.c > > to add a couple of commands to dynamically add and remove Skypiax > interfaces > > in a running FS process. However, this code does not replaces or > changes any > > previous code. Other then that there is no significant change in > > configuration steps. Though i did use mod_xml_curl to dynamically > update > > skypiax interface configuration in FS. > > > > > > Thank you. > > > > > > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli > > > wrote: > >> > >> Ciao Muhammad! > >> > >> What a good news! > >> > >> Centos is the most stable and performing platform for FS, so I > would > >> really love to test and document on the wiki how to have a stable > >> centos mod_skypiax installation. > >> > >> I'll find out your kernel ( Kernel > 2.6.18-92.1.22.el5.centos.plusPAE > >> ), and begin to test. In the mean time, do you have any hint, > special > >> procedure, etc you have done for having skypiax working well? > >> > >> Please, please, please let be in contact! :-) > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> > >> > >> > >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > >> Shahzad wrote: > >> > Sorry, i didn't visited the Jira link you mentioned. Now i know > the > >> > issue > >> > and I have replied it there. > >> > > >> > Thank you. > >> > > >> > > >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > >> > > >> > wrote: > >> >> > >> >> Hi all, > >> >> > >> >> there are problems for mod_skypiax in recent centos, with > more than a > >> >> handful of concurrent Skype calls. > >> >> > >> >> Probably the problem is ALSA-related. > >> >> > >> >> Until it is solved, for production please use Ubuntu 8.04 (see > below), > >> >> some other Linux distro (and please write here your > experience), or > >> >> Windows. > >> >> > >> >> I modified the wiki page to reflect this ( > >> >> http://wiki.freeswitch.org/wiki/ > Skypiax_Skype_Endpoint_and_Trunk ) > >> >> > >> >> If someone with CentOS knowledge can chime in I'll be > grateful :-). > >> >> > >> >> Please see Jira: http://jira.freeswitch.org/browse/ > MODSKYPIAX-34 for > >> >> all infos, and feel free to contact me directly. > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> ========================================= > >> >> www.celliax.org > >> >> via Pierlombardo 9, 20135 Milano > >> >> Italy > >> >> gmaruzz at celliax dot org > >> >> Cell : +39-347-2665618 > >> >> Fax : +39-02-87390039 > >> >> > >> >> _______________________________________________ > >> >> Freeswitch-users mailing list > >> >> Freeswitch-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Muhammad Shahzad > >> > ----------------------------------- > >> > CISCO Rich Media Communication Specialist (CRMCS) > >> > CISCO Certified Network Associate (CCNA) > >> > Cell: +92 334 422 40 88 > >> > MSN: shari_786pk at hotmail.com > >> > Email: shaheryarkh at googlemail.com > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/6419b0f6/attachment.html From gmaruzz at celliax.org Wed Jun 10 06:28:23 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 15:28:23 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: <7b197bef0906100628y25948a97x59af0ebd02c47d71@mail.gmail.com> On Wed, Jun 10, 2009 at 3:10 PM, dujinfang wrote: > btw, I have changed the start scripts a bit to start X and skype?separately, > glad to share it if someone?interested. I'm interested! :-) From gmaruzz at celliax.org Wed Jun 10 06:37:03 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 15:37:03 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> Message-ID: <7b197bef0906100637q7068ea56h666048867d968721@mail.gmail.com> On Wed, Jun 10, 2009 at 3:01 PM, Muhammad Shahzad wrote: > I am using this machine extensively for my development projects, including > Skypiax. Yesterday i gave a presentation to the board of directors of the > said firm, regarding existing status of my project. They tested the setup > with 2-3 concurrent SIP -> SKYPIAX and then SKYPIAX to SIP calls without any > problem. So, i believe this configuration works without any sound issue...! Until some times ago, I was used to test with 32 bit centos, and I've not seen problems (I tested with centos5.2). Now I'm installing 5.3 32 bit, and I'll report on my results On 64bit, on both centos5.2 and 5.3 I'm seeing problems with more than 6-10 concurrent calls, with less calls it was ok. BTW, it can be it is a problem with my hard disk (that is seen as an ide instead than as a SATA). I will check this too. I'll report back soon on the various issues... > > The second machine is my test machine in a remote data center. I didn't > prepare this machine, however, from SSH console i can see it has following > specs, > > 1. Intel(R) Xeon(R) CPU E5405? @ 2.00GHz with 4 GB of RAM. > 2. CentOS 5.3 with kernel 2.6.18-92.1.22.el5.centos.plusPAE. > 3. FS SVN revision Revision ID 13613. > > root ~# uname -a > Linux localhost.localdomain 2.6.18-92.1.22.el5.centos.plusPAE #1 SMP Wed Dec > 17 11:32:56 EST 2008 i686 i686 i386 GNU/Linux > > root ~# cat /etc/issue > CentOS release 5.3 (Final) > Kernel \r on an \m > > > Each machine that i use always, get update with yum update command BEFORE i > do anything else on it. > > Hope this info will be helpful for you. > > Can you give me step by step procedure of your testing that is producing > this bad sound result? I would like to perform this test on my both machines > and see if i get the same results too. > > Thank you. > > > On Wed, Jun 10, 2009 at 6:33 PM, Giovanni Maruzzelli > wrote: >> >> Ciao Muhammad, >> you're faster than light :-)! >> >> the patch will be integrated very soon, I'll let you know when I'm done >> it. >> >> Keep enhancements, patches, bug fixes, etc flowing! >> >> thanks again, and thanks to the firm that so quickly understood and >> authorized you, >> >> -giovanni >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 2:29 PM, Muhammad >> Shahzad wrote: >> > I am glad to share the patch to enable dynamic Skypiax interfaces in FS. >> > Please do note that however, that i started working on it on May 22, >> > 2009. >> > So any officaily changes made to mod_skypiax.c since then will not >> > appear in >> > it and will be lost if you apply this patch blindly. >> > >> > I request Giovanni Maruzzelli to carefully merge this patch in main >> > stream >> > code before committing it to FS SVN. >> > >> > Thank you. >> > >> > >> > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad >> > wrote: >> >> >> >> I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE >> >> enable >> >> kernel. I have heard 64bit ALSA drivers have bad sound issues, but >> >> never >> >> used it personally. >> >> >> >> As for source code of my modifications, i made those change to develop >> >> a >> >> customized commercial solution for large European firm, so i would need >> >> their permissions to provide you the required official patch. Let me >> >> write >> >> them an offical request for this. >> >> >> >> Thank you. >> >> >> >> >> >> On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli >> >> >> >> wrote: >> >>> >> >>> Ciao Muhammad, >> >>> >> >>> first thanks a lot for sharing your experience and help us in making a >> >>> better software! >> >>> >> >>> >From the name of the kernel, seems that you are using centos5.2 is >> >>> this >> >>> correct? >> >>> >> >>> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >> >>> >> >>> I'm now installing a centos5.2 (64), I will test it with centosplus >> >>> kernel and with its normal kernel. >> >>> >> >>> BTW, I would like *really* a lot to have and integrate your addition >> >>> to the code (also if it needs some labor from me, no problem). Would >> >>> you like to send it to me, so I will integrate in the main trunk and >> >>> you don't have no more to maintain it? (so you can develop other cool >> >>> features for mod_skypiax ;-) )? >> >>> >> >>> -giovanni >> >>> >> >>> Sincerely, >> >>> >> >>> Giovanni Maruzzelli >> >>> ========================================= >> >>> www.celliax.org >> >>> via Pierlombardo 9, 20135 Milano >> >>> Italy >> >>> gmaruzz at celliax dot org >> >>> Cell : +39-347-2665618 >> >>> Fax : +39-02-87390039 >> >>> >> >>> >> >>> >> >>> >> >>> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >> >>> Shahzad wrote: >> >>> > Thanks. I didn't make any special arrangements for FS or Skypiax to >> >>> > work on >> >>> > CentOS 5.3. I only enabled CentOS Plus yum repository and then >> >>> > install >> >>> > PAE >> >>> > kernel with following commands, >> >>> > >> >>> > root ~# yum update >> >>> > root ~# yum install kernel-PAE >> >>> > >> >>> > i installed PAE kernel just because i wanted to increase System RAM >> >>> > to >> >>> > 8GB >> >>> > before i deploy it for production use, so i can double or even >> >>> > triple >> >>> > Skypiax channels whenever i need so, without system or FS shutdown. >> >>> > >> >>> > As far as a skypiax configuration is concerned, i did modified >> >>> > mod_skypiax.c >> >>> > to add a couple of commands to dynamically add and remove Skypiax >> >>> > interfaces >> >>> > in a running FS process. However, this code does not replaces or >> >>> > changes any >> >>> > previous code. Other then that there is no significant change in >> >>> > configuration steps. Though i did use mod_xml_curl to dynamically >> >>> > update >> >>> > skypiax interface configuration in FS. >> >>> > >> >>> > >> >>> > Thank you. >> >>> > >> >>> > >> >>> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli >> >>> > >> >>> > wrote: >> >>> >> >> >>> >> Ciao Muhammad! >> >>> >> >> >>> >> What a good news! >> >>> >> >> >>> >> Centos is the most stable and performing platform for FS, so I >> >>> >> would >> >>> >> really love to test and document on the wiki how to have a stable >> >>> >> centos mod_skypiax installation. >> >>> >> >> >>> >> I'll find out your kernel ( Kernel >> >>> >> 2.6.18-92.1.22.el5.centos.plusPAE >> >>> >> ), and begin to test. In the mean time, do you have any hint, >> >>> >> special >> >>> >> procedure, etc you have done for having skypiax working well? >> >>> >> >> >>> >> Please, please, please let be in contact! :-) >> >>> >> >> >>> >> >> >>> >> Sincerely, >> >>> >> >> >>> >> Giovanni Maruzzelli >> >>> >> ========================================= >> >>> >> www.celliax.org >> >>> >> via Pierlombardo 9, 20135 Milano >> >>> >> Italy >> >>> >> gmaruzz at celliax dot org >> >>> >> Cell : +39-347-2665618 >> >>> >> Fax : +39-02-87390039 >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> >>> >> Shahzad wrote: >> >>> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know >> >>> >> > the >> >>> >> > issue >> >>> >> > and I have replied it there. >> >>> >> > >> >>> >> > Thank you. >> >>> >> > >> >>> >> > >> >>> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> >>> >> > >> >>> >> > wrote: >> >>> >> >> >> >>> >> >> Hi all, >> >>> >> >> >> >>> >> >> there are problems for mod_skypiax in recent centos, ?with more >> >>> >> >> than a >> >>> >> >> handful of concurrent Skype calls. >> >>> >> >> >> >>> >> >> Probably the problem is ALSA-related. >> >>> >> >> >> >>> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >> >>> >> >> below), >> >>> >> >> some other Linux distro (and please write here your experience), >> >>> >> >> or >> >>> >> >> Windows. >> >>> >> >> >> >>> >> >> I modified the wiki page to reflect this ( >> >>> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk >> >>> >> >> ) >> >>> >> >> >> >>> >> >> If someone with CentOS knowledge can chime in I'll be grateful >> >>> >> >> :-). >> >>> >> >> >> >>> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 >> >>> >> >> for >> >>> >> >> all infos, and feel free to contact me directly. >> >>> >> >> >> >>> >> >> -giovanni >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> Sincerely, >> >>> >> >> >> >>> >> >> Giovanni Maruzzelli >> >>> >> >> ========================================= >> >>> >> >> www.celliax.org >> >>> >> >> via Pierlombardo 9, 20135 Milano >> >>> >> >> Italy >> >>> >> >> gmaruzz at celliax dot org >> >>> >> >> Cell : +39-347-2665618 >> >>> >> >> Fax : +39-02-87390039 >> >>> >> >> >> >>> >> >> _______________________________________________ >> >>> >> >> Freeswitch-users mailing list >> >>> >> >> Freeswitch-users at lists.freeswitch.org >> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> >> http://www.freeswitch.org >> >>> >> > >> >>> >> > >> >>> >> > >> >>> >> > -- >> >>> >> > Muhammad Shahzad >> >>> >> > ----------------------------------- >> >>> >> > CISCO Rich Media Communication Specialist (CRMCS) >> >>> >> > CISCO Certified Network Associate (CCNA) >> >>> >> > Cell: +92 334 422 40 88 >> >>> >> > MSN: shari_786pk at hotmail.com >> >>> >> > Email: shaheryarkh at googlemail.com >> >>> >> > >> >>> >> > _______________________________________________ >> >>> >> > Freeswitch-users mailing list >> >>> >> > Freeswitch-users at lists.freeswitch.org >> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> > >> >>> >> > >> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> > http://www.freeswitch.org >> >>> >> > >> >>> >> > >> >>> >> >> >>> >> _______________________________________________ >> >>> >> Freeswitch-users mailing list >> >>> >> Freeswitch-users at lists.freeswitch.org >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >> >>> >> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> http://www.freeswitch.org >> >>> > >> >>> > >> >>> > >> >>> > -- >> >>> > Muhammad Shahzad >> >>> > ----------------------------------- >> >>> > CISCO Rich Media Communication Specialist (CRMCS) >> >>> > CISCO Certified Network Associate (CCNA) >> >>> > Cell: +92 334 422 40 88 >> >>> > MSN: shari_786pk at hotmail.com >> >>> > Email: shaheryarkh at googlemail.com >> >>> > >> >>> > _______________________________________________ >> >>> > Freeswitch-users mailing list >> >>> > Freeswitch-users at lists.freeswitch.org >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> > >> >>> > >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> > http://www.freeswitch.org >> >>> > >> >>> > >> >>> >> >>> _______________________________________________ >> >>> Freeswitch-dev mailing list >> >>> Freeswitch-dev at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> -- >> >> Muhammad Shahzad >> >> ----------------------------------- >> >> CISCO Rich Media Communication Specialist (CRMCS) >> >> CISCO Certified Network Associate (CCNA) >> >> Cell: +92 334 422 40 88 >> >> MSN: shari_786pk at hotmail.com >> >> Email: shaheryarkh at googlemail.com >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jingwei.yang at gmail.com Wed Jun 10 00:33:36 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Wed, 10 Jun 2009 15:33:36 +0800 Subject: [Freeswitch-users] What's the right way to use skypiax with dialplan Message-ID: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> Hi All, I just finished installing freeSwitch and Skypiax. And I'm able to use skype api directly via the sk command like the following: freeswitch at localhost.localdomain>sk console skypiax1 freeswitch at localhost.localdomain>sk CALL userAAA It works like a charm and userAAA is able to receive the call and answer it. However, I'm stuck in figuring out the right way to use Skypiax with a dialplan. I've put a dialplan like below into /usr/local/freeswitch/conf/dialplan/default.xml On the freeswitch console, I'm not sure how to trigger this dialplan correctly. I've tried something like freeswitch at localhost.localdomain>originate sofia/external/root at 192.168.1.1002909 API CALL [originate(sofia/external/root at 192.168.1.100 2909)] output: -ERR MANDATORY_IE_MISSING freeswitch at localhost.localdomain>originate sofia/external/localdomain at localhost 2909 API CALL [originate(sofia/external/localdomain at localhost 2909)] output: -ERR NORMAL_TEMPORARY_FAILURE All failed with errors indicated above. Please let me know what's the right way to originate the call. Thanks! Regards, -Jingwei p.s. my os is CentOS 5.3. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/a183818c/attachment.html From max.bridgewater at gmail.com Wed Jun 10 06:39:44 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 09:39:44 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls Message-ID: Hi, Getting to learn Freeswitch. So, please bear with me. Can somebody tells me how to do dial extension 1003 from the command line. I tried the following and the all failed originate 1003%192.168.10.103 originate 1003 at 192.168.10.103 originate 1003%192.168.10.103 & park() originate 1003 at 192.168.10.103 & park() Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/3171ea07/attachment.html From dujinfang at gmail.com Wed Jun 10 07:15:58 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 22:15:58 +0800 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100628y25948a97x59af0ebd02c47d71@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> <7b197bef0906100628y25948a97x59af0ebd02c47d71@mail.gmail.com> Message-ID: <49EB1F70-BD3D-4CF7-837F-6DE8AC01D5D5@gmail.com> On Jun 10, 2009, at 9:28 PM, Giovanni Maruzzelli wrote: > On Wed, Jun 10, 2009 at 3:10 PM, dujinfang wrote: >> btw, I have changed the start scripts a bit to start X and skype >> separately, >> glad to share it if someone interested. > > I'm interested! :-) > ok in jira. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Wed Jun 10 07:21:03 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 22:21:03 +0800 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: Message-ID: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> originate sofia/default/1003 &echo() originate user/1003 &echo() originate user/1003 &park() On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote: > Hi, > > Getting to learn Freeswitch. So, please bear with me. Can somebody > tells me how to do dial extension 1003 from the command line. > I tried the following and the all failed > > originate 1003%192.168.10.103 > originate 1003 at 192.168.10.103 > originate 1003%192.168.10.103 & park() > originate 1003 at 192.168.10.103 & park() > > Thanks, > Max. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/831d2410/attachment.html From dujinfang at gmail.com Wed Jun 10 07:25:43 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 22:25:43 +0800 Subject: [Freeswitch-users] What's the right way to use skypiax with dialplan In-Reply-To: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> References: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> Message-ID: <8210C83F-5ADD-4291-B807-AE2877EE3F6C@gmail.com> On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote: > Hi All, > > I just finished installing freeSwitch and Skypiax. And I'm able to > use skype api directly via the sk command like the following: > > freeswitch at localhost.localdomain>sk console skypiax1 > freeswitch at localhost.localdomain>sk CALL userAAA > > It works like a charm and userAAA is able to receive the call and > answer it. However, I'm stuck in figuring out the right way to use > Skypiax with a dialplan. I've put a dialplan like below into /usr/ > local/freeswitch/conf/dialplan/default.xml > > > > > > > > On the freeswitch console, I'm not sure how to trigger this dialplan > correctly. I've tried something like > > freeswitch at localhost.localdomain>originate sofia/external/root at 192.168.1.100 > 2909 > API CALL [originate(sofia/external/root at 192.168.1.100 2909)] output: > -ERR MANDATORY_IE_MISSING the problem is the dial string not the dialplan I think, why not try originate skypiax/ANY/userBBB 2909 it should call userBBB and bridge to userAAA. > > > freeswitch at localhost.localdomain>originate sofia/external/ > localdomain at localhost 2909 > API CALL [originate(sofia/external/localdomain at localhost 2909)] > output: > -ERR NORMAL_TEMPORARY_FAILURE > > > All failed with errors indicated above. Please let me know what's > the right way to originate the call. Thanks! > > Regards, > -Jingwei > > p.s. my os is CentOS 5.3. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/e5368126/attachment.html From larclap at yahoo.com Wed Jun 10 07:51:42 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 10 Jun 2009 07:51:42 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> Message-ID: <005301c9e9da$f8397ff0$e8ac7fd0$@com> There was no 0.0.0.0 anywhere. I used vi. I'll rotate the logs and restart FS without nc later today and report back. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 6:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/94abec9e/attachment-0001.html From max.bridgewater at gmail.com Wed Jun 10 07:54:41 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 10:54:41 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: Thanks, the first variant doesn't work for me. Any idea? I changed it to: originate sofia/internal/1003 at 192.168.10.103 & park() then i get the error message: "Cannot blind transfer 1 legged call". Also going through the wiki i discovered the concept of socket event and included the following extension under /usr/local/freeswitch/conf/dialplan/public/mysockt.xml: Can somebody gives me an example of dial string that would allow my call to be sent to this server socket? How would i place such a call using SJPhone? Best regards, Max. On Wed, Jun 10, 2009 at 10:21 AM, dujinfang wrote: > originate sofia/default/1003 &echo()originate user/1003 &echo() > originate user/1003 &park() > > > On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote: > > Hi, > > Getting to learn Freeswitch. So, please bear with me. Can somebody tells me > how to do dial extension 1003 from the command line. > I tried the following and the all failed > > originate 1003%192.168.10.103 > originate 1003 at 192.168.10.103 > originate 1003%192.168.10.103 & park() > originate 1003 at 192.168.10.103 & park() > > Thanks, > Max. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7c339943/attachment.html From mike at jerris.com Wed Jun 10 08:07:48 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Jun 2009 11:07:48 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: <36AC81BC-AFE5-4D38-901B-C293535C52CC@jerris.com> the default alias was removed from the default configs last week, so new configs don't have this anymore. On Jun 10, 2009, at 10:54 AM, Max Bridgewater wrote: > Thanks, > > the first variant doesn't work for me. Any idea? > I changed it to: > > originate sofia/internal/1003 at 192.168.10.103 & park() > > then i get the error message: "Cannot blind transfer 1 legged call". > > > > Also going through the wiki i discovered the concept of socket event > and included the following extension under /usr/local/freeswitch/ > conf/dialplan/public/mysockt.xml: > > > > break="on-true"> > > > > > > Can somebody gives me an example of dial string that would allow my > call to be sent to this server socket? How would i place such a call > using SJPhone? > > Best regards, > Max. > > On Wed, Jun 10, 2009 at 10:21 AM, dujinfang > wrote: > originate sofia/default/1003 &echo() > originate user/1003 &echo() > originate user/1003 &park() > > > On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote: >> Hi, >> >> Getting to learn Freeswitch. So, please bear with me. Can somebody >> tells me how to do dial extension 1003 from the command line. >> I tried the following and the all failed >> >> originate 1003%192.168.10.103 >> originate 1003 at 192.168.10.103 >> originate 1003%192.168.10.103 & park() >> originate 1003 at 192.168.10.103 & park() >> >> Thanks, >> Max. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7d80b76b/attachment.html From brian at freeswitch.org Wed Jun 10 08:11:25 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 10:11:25 -0500 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings If you're calling a locally registered user... you need to use user/ user at domain which uses dial-string from the params on the user or directory. Or sofia/profile/user%domain /b On Jun 10, 2009, at 9:54 AM, Max Bridgewater wrote: > Thanks, > > the first variant doesn't work for me. Any idea? > I changed it to: > > originate sofia/internal/1003 at 192.168.10.103 & park() > > then i get the error message: "Cannot blind transfer 1 legged call". > > > > Also going through the wiki i discovered the concept of socket event > and included the following extension under /usr/local/freeswitch/ > conf/dialplan/public/mysockt.xml: > > > > break="on-true"> > > > > > > Can somebody gives me an example of dial string that would allow my > call to be sent to this server socket? How would i place such a call > using SJPhone? > > Best regards, > Max. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/1b2a2f1a/attachment.html From max.bridgewater at gmail.com Wed Jun 10 09:18:49 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 12:18:49 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: Thanks Folks; I'm making progress. The following origination string does make my non-registered SJPhone ring: {origination_caller_id_number=2000}sofia/external/some at 192.168.50.67 But why isn't it caught by the following extension? On Wed, Jun 10, 2009 at 11:11 AM, Brian West wrote: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > If you're calling a locally registered user... you need to use > user/user at domain which uses dial-string from the params on the user or > directory. Or sofia/profile/user%domain > > /b > > > On Jun 10, 2009, at 9:54 AM, Max Bridgewater wrote: > > Thanks, > > the first variant doesn't work for me. Any idea? > I changed it to: > > originate sofia/internal/1003 at 192.168.10.103 & park() > > then i get the error message: "Cannot blind transfer 1 legged call". > > > > Also going through the wiki i discovered the concept of socket event and > included the following extension under > /usr/local/freeswitch/conf/dialplan/public/mysockt.xml: > > > > break="on-true"> > /> > > > > > Can somebody gives me an example of dial string that would allow my call to > be sent to this server socket? How would i place such a call using SJPhone? > > Best regards, > Max. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7d58a043/attachment-0001.html From mike at jerris.com Wed Jun 10 09:52:51 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Jun 2009 12:52:51 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Dialplan_XML break="on-true" ? On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote: > Thanks Folks; I'm making progress. The following origination string > does make my non-registered SJPhone ring: > > {origination_caller_id_number=2000}sofia/external/some at 192.168.50.67 > > > But why isn't it caught by the following extension? > > > > > break="on-true"> > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/69f988cc/attachment.html From paul at ringcarrier.com Wed Jun 10 10:02:16 2009 From: paul at ringcarrier.com (Paul Mahler) Date: Wed, 10 Jun 2009 10:02:16 -0700 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? Message-ID: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Hello Everybody, I have a large project coming up. I'm interested in using Freeswitch instead of SER and Asterisk. What is the current status of Freeswitch? Can I safely use it in a large scale commercial environment? How active is the Freeswitch developer community? I am concerned that Freeswitch doesn't seem to have gained much traction compared to Asterisk. How viable is the Freeswitch project? Please help me understand why I can safely shit can Asterisk and move to Freeswitch. Thank You, Paul _________________ Paul Mahler paul at ringcarrier.com From max.bridgewater at gmail.com Wed Jun 10 10:20:01 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 13:20:01 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: Well, i assume break="on-true" means, that if this extension is matched then execute its actions and stop there. That would correspond to what i'm trying to do. Anyway i removed this attribute and still nothing is being sent to the socket. Let me give you more context. This extension is put in a separate file AS IS under /usr/local/freeswitch/conf/dialplan/public/myextension.xml. Is this the right place? I have the impression that it is not seen or processed by Freeswitch. The latest thing i tried is to wrap the extension in a context element with name attribute "public". It did'nt help. Any clue? On Wed, Jun 10, 2009 at 12:52 PM, Michael Jerris wrote: > http://wiki.freeswitch.org/wiki/Dialplan_XML > break="on-true" ? > > On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote: > > Thanks Folks; I'm making progress. The following origination string does > make my non-registered SJPhone ring: > > {origination_caller_id_number=2000}sofia/external/some at 192.168.50.67 > > > But why isn't it caught by the following extension? > > > > > break="on-true"> > /> > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/b81b14cb/attachment.html From benj at teliax.com Wed Jun 10 10:18:43 2009 From: benj at teliax.com (Ben Jones) Date: Wed, 10 Jun 2009 11:18:43 -0600 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Message-ID: <4A2FEAF3.1080807@teliax.com> Hi Paul, I'll tell you this. We (Teliax) made the decision to use FreeSWITCH instead of Asterisk in late 2008 and we haven't looked back since. We're a fairly large SIP/IAX provider with four POPs located throughout the US, each one running FS. The dev community surrounding FS is excellent, they're always helpful and if you check the changelogs (http://fisheye.freeswitch.org/changelog/FreeSWITCH/) you can clearly see there's all sorts of work going on daily. As a company with 5000+ customers with an emphasis on call completion and high quality, we're happy to say that we can rely on FreeSWITCH every single day to provide us with what Asterisk couldn't. Don't get me wrong, Asterisk is a good PBX (it's quick to deploy, relatively easy, and well supported) but try pushing thousands of calls through it at once without it choking. FreeSWITCH handles it with 94% processor idle. I realize I might not answer all your questions, but I just wanted to throw in our vote for you to replace Asterisk with FreeSWITCH. Maybe seeing a real-world example such as ours will help your decision to make the switch. (pun fully intended) Hope this helps, Ben J Paul Mahler wrote: > Hello Everybody, > > I have a large project coming up. I'm interested in using Freeswitch > instead of SER and Asterisk. > > What is the current status of Freeswitch? Can I safely use it in a > large scale commercial environment? How active is the Freeswitch > developer community? > > I am concerned that Freeswitch doesn't seem to have gained much > traction compared to Asterisk. How viable is the Freeswitch project? > > Please help me understand why I can safely shit can Asterisk and move > to Freeswitch. > > Thank You, > > Paul > _________________ > Paul Mahler > paul at ringcarrier.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From helmut.kuper at ewetel.de Wed Jun 10 10:32:41 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 10 Jun 2009 19:32:41 +0200 Subject: [Freeswitch-users] Database and "Too many open files" Problem Message-ID: <4A2FEE39.1030709@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I updated to latest trunk. Everything compiled well. FS starts up without errors. Calls are executed successfully. But after a few minutes with very few call activity I get this lines on console: 2009-06-10 18:21:16.52527 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) 2009-06-10 18:21:16.152501 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) 2009-06-10 18:21:16.252481 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) ... 2009-06-10 18:22:37.121405 [CRIT] switch_core_sqldb.c:209 SQL thread unable to commit transaction, records lost! I modified switch_core_sqldb.c to output the actual sql statement as well to get an idea which database is affected. It's the core-db. Mixed with that lines above I got lines like this: 2009-06-10 18:21:32.656605 [ERR] mod_sndfile.c:194 Error Opening File [/opt/app/voip/ippbx/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav] [System error : Too many open files in system.] This may points to the ulimit thing, cause I run FS as non root (on centos 5.3). ulimit -a shows: [ippbx at ippbx-test-node0 ~]$ ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited /etc/security/limits.conf contains: ippbx - nofile 999999 ippbx - core unlimited ippbx - data unlimited ippbx - fsize unlimited ippbx - sigpending unlimited ippbx - msgqueue unlimited ippbx - nproc unlimited ippbx - locks unlimited So has anyone an idea what's wrong? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKL+454tZeNddg3dwRAqcCAJsFSEWuI2X1fvOF2rmIFoSmRdBZzQCggD8B MQiS7RZkOoL9lhV8LV+pL7s= =v7QY -----END PGP SIGNATURE----- From benj at teliax.com Wed Jun 10 10:34:04 2009 From: benj at teliax.com (Ben Jones) Date: Wed, 10 Jun 2009 11:34:04 -0600 Subject: [Freeswitch-users] Clarification on start_dtmf_generate? Message-ID: <4A2FEE8C.8090906@teliax.com> Hello all, The documentation has this to say on start_dtmf_generate: As an example, Placing this in the Dialplan prior to bridging a call, will allow a phone set to rfc2833 ( info ) to send DTMF tones (in-band ) out to the recipient (IVR) or Auto Attendants. Thus changing the outgoing routing from (info) to (in-band). From what I understand rfc2833 != info. Does this mean if a user is set for rfc2833 OR info that FS will generate inband tones to send out? Does it only generate tones for info? If I can get any clarification on this I'll be more than happy to update the documentation. Thanks, Ben J -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From brian at freeswitch.org Wed Jun 10 10:40:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 12:40:58 -0500 Subject: [Freeswitch-users] Clarification on start_dtmf_generate? In-Reply-To: <4A2FEE8C.8090906@teliax.com> References: <4A2FEE8C.8090906@teliax.com> Message-ID: <72CBB468-4382-4F09-B1F7-9CC678F61243@freeswitch.org> Yes. /b On Jun 10, 2009, at 12:34 PM, Ben Jones wrote: > Does this mean if a user is set for rfc2833 OR info that FS will > generate inband tones to send out? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/5e5c8d3b/attachment-0001.html From freeswitch-users at digitaldan.com Wed Jun 10 10:57:16 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 10 Jun 2009 11:57:16 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <87f2f3b90906091506s5da8b925u256eaeaaf8c0fbb3@mail.gmail.com> Message-ID: <9943583.12931244656630422.JavaMail.daniel@radio> Yep, we tracked it down to the originating pbx ( a cisco call manager) which had a 12 hour limit on outbound calls, thanks for your help. D- ----- Original Message ----- From: "Michael Collins" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 9, 2009 4:06:01 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording time limit? On Tue, Jun 9, 2009 at 2:55 PM, Brian West < brian at freeswitch.org > wrote: Someone hung the call up thats all I can imagine. Its not going to be us but maybe the far side you're talking to. Look at the sip trace bet the far end is sending you a BYE This would make sense. Some systems will have a hard limit on the length of a call and will disconnect automatically because they interpret a 12 hour call as a "problem" and the "solution" is to hang up. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/df7f1bd8/attachment.html From msc at freeswitch.org Wed Jun 10 11:04:37 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 10 Jun 2009 11:04:37 -0700 Subject: [Freeswitch-users] Clarification on start_dtmf_generate? In-Reply-To: <72CBB468-4382-4F09-B1F7-9CC678F61243@freeswitch.org> References: <4A2FEE8C.8090906@teliax.com> <72CBB468-4382-4F09-B1F7-9CC678F61243@freeswitch.org> Message-ID: I will clarify the point on the wiki since it is a bit inaccurate -MC Sent from my iPhone On Jun 10, 2009, at 10:40 AM, Brian West wrote: > Yes. > > /b > > On Jun 10, 2009, at 12:34 PM, Ben Jones wrote: > >> Does this mean if a user is set for rfc2833 OR info that FS will >> generate inband tones to send out? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/bf357658/attachment.html From nik.middleton at noblesolutions.co.uk Wed Jun 10 11:30:55 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 10 Jun 2009 19:30:55 +0100 Subject: [Freeswitch-users] Problems with make current Message-ID: Hi Guys, Ran make current today, and am getting the following errors. I ran bootstrap and configure, but still get these messages. Any ideas ? Looks like I'm now missing some libraries Regards, configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/f9ec3533/attachment.html From brian at freeswitch.org Wed Jun 10 11:36:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 13:36:01 -0500 Subject: [Freeswitch-users] Problems with make current In-Reply-To: References: Message-ID: <3C17ED7A-A837-45D3-8469-3CDDABF9EB83@freeswitch.org> need to rebootstrap. /b On Jun 10, 2009, at 1:30 PM, Nik Middleton wrote: > Hi Guys, > > Ran make current today, and am getting the following errors. I ran > bootstrap and configure, but still get these messages. > > Any ideas ? Looks like I?m now missing some libraries > > Regards, > > configure: configuring in libs/pcre > configure: running /bin/sh './configure.gnu' --prefix=/usr/local/ > freeswitch --cache-file=/dev/null --srcdir=. > configure: error: cannot find sources (pcre.h.in) in . > configure: error: /bin/sh './configure.gnu' failed for libs/pcre Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/020f5c63/attachment.html From mike at jerris.com Wed Jun 10 11:50:07 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Jun 2009 14:50:07 -0400 Subject: [Freeswitch-users] Problems with make current In-Reply-To: References: Message-ID: <9B3365B3-9C75-412D-B3CF-623F32F26B8C@jerris.com> your svn update failed, rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make current On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote: > Hi Guys, > > Ran make current today, and am getting the following errors. I ran > bootstrap and configure, but still get these messages. > > Any ideas ? Looks like I?m now missing some libraries > > Regards, > > configure: configuring in libs/pcre > configure: running /bin/sh './configure.gnu' --prefix=/usr/local/ > freeswitch --cache-file=/dev/null --srcdir=. > configure: error: cannot find sources (pcre.h.in) in . > configure: error: /bin/sh './configure.gnu' failed for libs/pcre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/5fcb649c/attachment-0001.html From marketing at cluecon.com Wed Jun 10 13:11:53 2009 From: marketing at cluecon.com (Michael Collins) Date: Wed, 10 Jun 2009 13:11:53 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Important Reminders Message-ID: <87f2f3b90906101311g4e41c324p279918f855703e6b@mail.gmail.com> Friends, ClueCon 2009 is fast approaching! We are definitely looking forward to seeing everyone in Chicago this August. If you haven't finalized your plans to attend, please do so right away. Time is running out! The early-bird registration special of $499 per person will expire at the end of June - only 20 days away. Also, registration as a whole ends on July 21st, so don't delay. Lastly, be sure to book your room at the Chicago Wyndham. Try expedia.com to see what kind of deals are still available. Please call 877.742.CLUE to get registered today! -The ClueCon Team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/33cd63ad/attachment.html From nik.middleton at noblesolutions.co.uk Wed Jun 10 13:53:31 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 10 Jun 2009 21:53:31 +0100 Subject: [Freeswitch-users] Problems with make current In-Reply-To: <9B3365B3-9C75-412D-B3CF-623F32F26B8C@jerris.com> References: <9B3365B3-9C75-412D-B3CF-623F32F26B8C@jerris.com> Message-ID: Thanks, that did the trick Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 10 June 2009 19:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with make current your svn update failed, rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make current On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote: Hi Guys, Ran make current today, and am getting the following errors. I ran bootstrap and configure, but still get these messages. Any ideas ? Looks like I'm now missing some libraries Regards, configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/d1c14146/attachment.html From larclap at yahoo.com Wed Jun 10 14:04:08 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 10 Jun 2009 14:04:08 -0700 Subject: [Freeswitch-users] Remove example.com gateway? Message-ID: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> Is it OK to remove the example.com gateway? I removed the example.xml files in sip_profiles/external and sip_profiles/internal and changed the default_provider from example.com to myprovider.com. But I still see myprovider.com as a gateway in sofia status. How do I get rid of this, of course, if it's OK. I want to simplify the configuration. I dialed 18885551212 and it went into [local.example.com]; and 8885551212 it went into [eavesdrop]. I think I'm talking more than just the gateway, but I do want to make the dialplan work in the way I expect. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/f2cef6bd/attachment.html From brian at freeswitch.org Wed Jun 10 14:10:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 16:10:01 -0500 Subject: [Freeswitch-users] Remove example.com gateway? In-Reply-To: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> References: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> Message-ID: You can remove it all you want but those settings in vars.xml won't do anything because its expanded in conf/directory/default/example.com.xml /b On Jun 10, 2009, at 4:04 PM, Lars Zeb wrote: > Is it OK to remove the example.com gateway? I removed the > example.xml files in sip_profiles/external and sip_profiles/internal > and changed the default_provider from example.com to myprovider.com. > > But I still see myprovider.com as a gateway in sofia status. How do > I get rid of this, of course, if it?s OK. > > I want to simplify the configuration. I dialed 18885551212 and it > went into [local.example.com]; and 8885551212 it went into > [eavesdrop]. I think I?m talking more than just the gateway, but I > do want to make the dialplan work in the way I expect. > > Thanks, Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/96a513d8/attachment.html From anthony.minessale at gmail.com Wed Jun 10 14:14:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 10 Jun 2009 16:14:26 -0500 Subject: [Freeswitch-users] Database and "Too many open files" Problem In-Reply-To: <4A2FEE39.1030709@ewetel.de> References: <4A2FEE39.1030709@ewetel.de> Message-ID: <191c3a030906101414j1107e49bk3aa8fb778454451c@mail.gmail.com> This is a bug in the latest wanpipe. you can wait for the next release of wanpipe/openzap or you can downgrade to wanpipe 4.1 and rebuild FS On Wed, Jun 10, 2009 at 12:32 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > today I updated to latest trunk. Everything compiled well. FS starts up > without errors. Calls are executed successfully. > > But after a few minutes with very few call activity I get this lines on > console: > > > 2009-06-10 18:21:16.52527 [ERR] switch_core_sqldb.c:95 SQL ERR [unable > to open database file] (update tasks set task_sql_manager='' where > task_id=1; > ) > 2009-06-10 18:21:16.152501 [ERR] switch_core_sqldb.c:95 SQL ERR [unable > to open database file] (update tasks set task_sql_manager='' where > task_id=1; > ) > 2009-06-10 18:21:16.252481 [ERR] switch_core_sqldb.c:95 SQL ERR [unable > to open database file] (update tasks set task_sql_manager='' where > task_id=1; > ) > > ... > > 2009-06-10 18:22:37.121405 [CRIT] switch_core_sqldb.c:209 SQL thread > unable to commit transaction, records lost! > > I modified switch_core_sqldb.c to output the actual sql statement as > well to get an idea which database is affected. It's the core-db. > > > > Mixed with that lines above I got lines like this: > > 2009-06-10 18:21:32.656605 [ERR] mod_sndfile.c:194 Error Opening File > > [/opt/app/voip/ippbx/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav] > [System error : Too many open files in system.] > > > This may points to the ulimit thing, cause I run FS as non root (on > centos 5.3). > > ulimit -a shows: > [ippbx at ippbx-test-node0 ~]$ ulimit -a > core file size (blocks, -c) 0 > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) 32 > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 10240 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > /etc/security/limits.conf contains: > ippbx - nofile 999999 > ippbx - core unlimited > ippbx - data unlimited > ippbx - fsize unlimited > ippbx - sigpending unlimited > ippbx - msgqueue unlimited > ippbx - nproc unlimited > ippbx - locks unlimited > > > So has anyone an idea what's wrong? > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKL+454tZeNddg3dwRAqcCAJsFSEWuI2X1fvOF2rmIFoSmRdBZzQCggD8B > MQiS7RZkOoL9lhV8LV+pL7s= > =v7QY > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/a8983233/attachment-0001.html From msc at freeswitch.org Wed Jun 10 14:51:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Jun 2009 14:51:00 -0700 Subject: [Freeswitch-users] Remove example.com gateway? In-Reply-To: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> References: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> Message-ID: <87f2f3b90906101451p3af74a65ta52da2646954500a@mail.gmail.com> On Wed, Jun 10, 2009 at 2:04 PM, Lars Zeb wrote: > Is it OK to remove the example.com gateway? I removed the example.xml > files in sip_profiles/external and sip_profiles/internal and changed the > default_provider from example.com to myprovider.com. > > If you just want it to "go away" then you can always rename the file to something like 'example.com.noload' -MC > > > But I still see myprovider.com as a gateway in sofia status. How do I get > rid of this, of course, if it?s OK. > > > > I want to simplify the configuration. I dialed 18885551212 and it went into > [local.example.com]; and 8885551212 it went into [eavesdrop]. I think I?m > talking more than just the gateway, but I do want to make the dialplan work > in the way I expect. > > > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/2bc635a4/attachment.html From msc at freeswitch.org Wed Jun 10 15:34:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Jun 2009 15:34:52 -0700 Subject: [Freeswitch-users] A Few New Blog Posts Message-ID: <87f2f3b90906101534q52d4ad06l9524fc57c10a9407@mail.gmail.com> FYI, I've added a few new posts on the main FreeSWITCH page: http://www.freeswitch.org/node/190 - OpenSimulator on EC2 http://www.freeswitch.org/node/191 - Rob Smart FS as-home-PBX (U.K.) how-to Enjoy! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/798e0975/attachment.html From palani.sivagurunathan at gmail.com Wed Jun 10 15:45:08 2009 From: palani.sivagurunathan at gmail.com (palani vel) Date: Wed, 10 Jun 2009 18:45:08 -0400 Subject: [Freeswitch-users] palani vel sent you a Friend Request on Yaari Message-ID: palani vel wants you to join Yaari! Is palani your friend? Yes, palani is my friend! No, palani isn't my friend. Please respond or palani may think you said no :( Thanks, The Yaari Team ----------------------------------------------------------- Yaari Inc., 358 Angier Ave NE Atlanta, GA 30312 Privacy Policy | Unsubscribe | Terms of Service YaariNYX927AYT966XXO323IQT265 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/0f0eec8f/attachment.html From larclap at yahoo.com Wed Jun 10 16:54:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 10 Jun 2009 16:54:38 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> Message-ID: <015b01c9ea26$d0abb2e0$720318a0$@com> Rupa, I think the console log has information in it that log/freeswitch.log does not. Console: [root at fs bin]# ./freeswitch 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (Invalid or incomplete multibyte or wide character) 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! log/freeswitch.log: 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock interface 'console' to wait for existing references. (from previous Freeswitch invocation) 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding Dialplan 'enum' 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding Application 'enum' 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum' 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum_auto' 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default template. 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template example. 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template linksys. 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template asterisk. I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console log. However, the disk log begins at 16:12:58, whereas the console log starts at 16:12:50. The console log finishes its NAT and UPnP reporting before the disk log begins, so I wouldn't see any 0.0.0.0 if it were present. The [ERR] was due to me removing example.xml from sip_profiles/internal. I put it back after this. I don't understand the following command in conf/sofia.conf.xml. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 6:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/5d6b3763/attachment-0001.html From msc at freeswitch.org Wed Jun 10 17:18:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Jun 2009 17:18:10 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015b01c9ea26$d0abb2e0$720318a0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> <015b01c9ea26$d0abb2e0$720318a0$@com> Message-ID: <87f2f3b90906101718g72911b8fv326ee9c470fdf097@mail.gmail.com> On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb wrote: > Rupa, > > > > I think the console log has information in it that log/freeswitch.log does > not. > > > > Console: > > [root at fs bin]# ../freeswitch > > 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing > Engine. > > 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch > thread 0 > > 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (No such file or directory) > > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (Invalid or incomplete multibyte or wide character) > > 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT > > 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 > > 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 > > 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 > > 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 > > 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 > > 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP > > 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT > detected! > > > > log/freeswitch.log: > > 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock > interface 'console' to wait for existing references. (from previous > Freeswitch invocation) > > 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding > Dialplan 'enum' > > 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding > Application 'enum' > > 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum' > > 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum_auto' > > 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default > template. > > 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. > > 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. > > 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template > example. > > 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. > > 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template > linksys. > > 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template > asterisk. > > > > I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console > log. However, the disk log begins at 16:12:58, whereas the console log > starts at 16:12:50. The console log finishes its NAT and UPnP reporting > before the disk log begins, so I wouldn?t see any 0.0.0.0 if it were > present. > > > > The [ERR] was due to me removing example.xml from sip_profiles/internal. I > put it back after this. I don?t understand the following command in > conf/sofia.conf.xml. > > > > > > I think this is just a cosmetic error. You could probably put an empty xml file in sip_profiles/internal and be done with it. Or possibly have just an empty include node, like "" Try it out and report back - we're dying to know what happens! ;) -MC > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 6:05 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > if you haven't changed your logging, then it is probably ok. The 0.0.0.0 > thing is logged at error level, so will show up in the logs. How did you > search? Grep? > > grep '0\.0\.0\.0' freeswitch.log > > On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: > > Rupa, > > > > What options do I have for setting up logging? I?m sorry, but I don?t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 5:19 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > > > On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > > You might want to review how you have your logging setup then. The example > I gave you was copied/pasted out of my freeswitch.log file while testing > this fix. > > > -- > -Rupa > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/245ee933/attachment.html From john at feith.com Wed Jun 10 17:31:53 2009 From: john at feith.com (John Wehle) Date: Wed, 10 Jun 2009 20:31:53 -0400 (EDT) Subject: [Freeswitch-users] Finding all active calls belonging to the same phone Message-ID: <200906110031.n5B0Vr7S006625@jwlab.FEITH.COM> To duplicate our old PBX park functionality I need for a user who's on a call to be able to pick up a second line and dial a number to park the other call which is on his phone. I have something working, however am curious if there's a better way to accomplish this. Specifically I'm curious if there's a recommended way to find all the calls to / from the same phone / channel. I ended up configuring *5 to call a javascript program which: a) Gets the uuid from the session and uses it to search "show channels" to find the channel name. b) Normalizes the channel name and uses it to search "show channels" to find an uuid associated with the channel which is different from the one that invoked *5. c) Uses the uuid from "b" to search "show calls" to find the peer uuid. d) Uses uuid_setvar to set hangup_after_bridge=false and uuid_transfer to transfer the peer uuid to the proper fifo. One of the problems I ran into is the channel name has slightly different formats depending on whether it is an inbound or outbound channel. E.g.: sofia/internal/1003 at XXX.XXX.XXX.XXX sofia/internal/1003 at XXX.XXX.XXX.XXX:5060 sofia/internal/sip:1003 at YYY.YYY.YYY.YYY:5060;transport=udp;... where XXX is the freeswitch box and YYY is the phone. I created the following function to normalize the channel name for comparison: function normalize_channel_name (name, direction, ip_addr) { var re = /^sofia\//g; var length = name.search (re); var new_name = name; if (length == -1) return new_name; if (direction == "inbound") { re = /@.*$/g; new_name = name.replace (re, "@" + ip_addr); } else if (direction == "outbound") { re = /\/sip:(.*@[^:]*):.*$/g; new_name = name.replace (re, "/$1"); } return new_name; } Suggestions for a better approach? Keep in mind that my existing user population expects (for better or worse) to use *5 to park the call on their phone so I'm somewhat limited in what I can do. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From jcromes at gmail.com Wed Jun 10 17:49:01 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Wed, 10 Jun 2009 19:49:01 -0500 Subject: [Freeswitch-users] Web page thoughts Message-ID: <4A30547D.4090101@gmail.com> My company is currently investigating a couple of projects that may take me in the direction of FreeSwitch... In general, our management does not often consider open source software for projects such as this, but I've been successful in proving to them recently that open source can deliver. FreeSwitch is a *very* professional and polished product, I can tell - from the code and from the community. Unfortunately, I've been hesitant to send people to your webpage lately because it went downhill a few weeks ago. Whenever I think about one of our executives going to your webpage (after my recommendation) and seeing a picture of people clanking beer glasses, or some idiot tied up in phone cables, I cringe. I know you're advertising for ClueCon, but honestly, some of those huge images on your front page really knock your product down a peg in professionalism. Anyway, I'm pretty new to the community and I don't claim to be a web designer. You have an excellent piece of software, but if I didn't already know that about FreeSwitch, your webpage would not make a good first impression. Please take that for what it's worth... I wanted to voice my opinion because if I'm thinking it, others may be as well. Thoughts anyone? J (Have I mentioned how awesome your source browser is though??!!) From rupa at rupa.com Wed Jun 10 17:58:30 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 10 Jun 2009 19:58:30 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015b01c9ea26$d0abb2e0$720318a0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> <015b01c9ea26$d0abb2e0$720318a0$@com> Message-ID: Oh, it is probable that logging is initialized after nat.... so the first pass won't show up in the filesystem logs. It would should up on subsequent nat initialization (what I was testing with some new code). On Wed, Jun 10, 2009 at 6:54 PM, Lars Zeb wrote: > Rupa, > > > > I think the console log has information in it that log/freeswitch.log does > not. > > > > Console: > > [root at fs bin]# ../freeswitch > > 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing > Engine. > > 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch > thread 0 > > 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (No such file or directory) > > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (Invalid or incomplete multibyte or wide character) > > 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT > > 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 > > 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 > > 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 > > 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 > > 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 > > 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP > > 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT > detected! > > > > log/freeswitch.log: > > 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock > interface 'console' to wait for existing references. (from previous > Freeswitch invocation) > > 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding > Dialplan 'enum' > > 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding > Application 'enum' > > 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum' > > 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum_auto' > > 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default > template. > > 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. > > 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. > > 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template > example. > > 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. > > 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template > linksys. > > 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template > asterisk. > > > > I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console > log. However, the disk log begins at 16:12:58, whereas the console log > starts at 16:12:50. The console log finishes its NAT and UPnP reporting > before the disk log begins, so I wouldn?t see any 0.0.0.0 if it were > present. > > > > The [ERR] was due to me removing example.xml from sip_profiles/internal. I > put it back after this. I don?t understand the following command in > conf/sofia.conf.xml. > > > > > > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 6:05 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > if you haven't changed your logging, then it is probably ok. The 0.0.0.0 > thing is logged at error level, so will show up in the logs. How did you > search? Grep? > > grep '0\.0\.0\.0' freeswitch.log > > On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: > > Rupa, > > > > What options do I have for setting up logging? I?m sorry, but I don?t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 5:19 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > > > On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > > You might want to review how you have your logging setup then. The example > I gave you was copied/pasted out of my freeswitch.log file while testing > this fix. > > > -- > -Rupa > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/88df1ec5/attachment-0001.html From jingwei.yang at gmail.com Wed Jun 10 18:25:21 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 11 Jun 2009 09:25:21 +0800 Subject: [Freeswitch-users] What's the right way to use skypiax with dialplan In-Reply-To: <8210C83F-5ADD-4291-B807-AE2877EE3F6C@gmail.com> References: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> <8210C83F-5ADD-4291-B807-AE2877EE3F6C@gmail.com> Message-ID: <13529f9d0906101825y76be738bhf957e8b30d829b94@mail.gmail.com> Yes, it's the right way to go! Thanks, man! On Wed, Jun 10, 2009 at 10:25 PM, dujinfang wrote: > > On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote: > > Hi All, > > I just finished installing freeSwitch and Skypiax. And I'm able to use > skype api directly via the sk command like the following: > > freeswitch at localhost.localdomain>sk console skypiax1 > freeswitch at localhost.localdomain>sk CALL userAAA > > It works like a charm and userAAA is able to receive the call and answer > it. However, I'm stuck in figuring out the right way to use Skypiax with a > dialplan. I've put a dialplan like below into > /usr/local/freeswitch/conf/dialplan/default.xml > > > > > > > > On the freeswitch console, I'm not sure how to trigger this dialplan > correctly. I've tried something like > > freeswitch at localhost.localdomain>originate sofia/external/ > root at 192.168.1.100 2909 > API CALL [originate(sofia/external/root at 192.168.1.100 2909)] output: > -ERR MANDATORY_IE_MISSING > > > the problem is the dial string not the dialplan I think, why not try > > originate skypiax/ANY/userBBB 2909 > > it should call userBBB and bridge to userAAA. > > > > freeswitch at localhost.localdomain>originate > sofia/external/localdomain at localhost 2909 > API CALL [originate(sofia/external/localdomain at localhost 2909)] output: > -ERR NORMAL_TEMPORARY_FAILURE > > > All failed with errors indicated above. Please let me know what's the right > way to originate the call. Thanks! > > Regards, > -Jingwei > > p.s. my os is CentOS 5.3. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/f1c89438/attachment.html From diego.viola at gmail.com Wed Jun 10 18:39:24 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 21:39:24 -0400 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <20090609033051.GA28848@jdc.jasonjgw.net> References: <005001c9e8a9$76c5c780$64515680$@com> <20090609033051.GA28848@jdc.jasonjgw.net> Message-ID: <86a32abc0906101839n67119d0fs935401a4a45e5f32@mail.gmail.com> Another vote for Git here :). On Mon, Jun 8, 2009 at 11:30 PM, Jason White wrote: > Lars Zeb wrote: > > I had a working FS installation which I messed up by doing a fresh > install. > > I tried to integrate all my custom changes, but I'm sure I screwed > something > > up. > > Git is an excellent tool for keeping track of FreeSWITCH configuration > changes. The history of my configuration is maintained in a git repository > under /opt/freeswitch/conf - git simply creates a .git subdirectory to > store > all of the revisions as they are committed. > > Git revert and git stash have been very useful at times, not to mention git > reset --hard. > > Since Git is used for Linux kernel development, it should be available from > most recent Linux distributions, and it can probably be compiled for other > Unix-like environments as well. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/1819885a/attachment.html From brian at freeswitch.org Wed Jun 10 19:29:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 21:29:49 -0500 Subject: [Freeswitch-users] Finding all active calls belonging to the same phone In-Reply-To: <200906110031.n5B0Vr7S006625@jwlab.FEITH.COM> References: <200906110031.n5B0Vr7S006625@jwlab.FEITH.COM> Message-ID: Why not use the presence events to keep that state? /b On Jun 10, 2009, at 7:31 PM, John Wehle wrote: > To duplicate our old PBX park functionality I need for a user who's > on a call to be able to pick up a second line and dial a number to > park the other call which is on his phone. I have something working, > however am curious if there's a better way to accomplish this. > > Specifically I'm curious if there's a recommended way to find all the > calls to / from the same phone / channel. > > I ended up configuring *5 to call a javascript program which: > > a) Gets the uuid from the session and uses it to search "show > channels" > to find the channel name. > > b) Normalizes the channel name and uses it to search "show channels" > to find an uuid associated with the channel which is different > from > the one that invoked *5. > > c) Uses the uuid from "b" to search "show calls" to find the peer > uuid. > > d) Uses uuid_setvar to set hangup_after_bridge=false and > uuid_transfer > to transfer the peer uuid to the proper fifo. > > One of the problems I ran into is the channel name has slightly > different > formats depending on whether it is an inbound or outbound channel. > E.g.: > > sofia/internal/1003 at XXX.XXX.XXX.XXX > sofia/internal/1003 at XXX.XXX.XXX.XXX:5060 > sofia/internal/sip:1003 at YYY.YYY.YYY.YYY:5060;transport=udp;... > > where XXX is the freeswitch box and YYY is the phone. I created the > following function to normalize the channel name for comparison: > > function normalize_channel_name (name, direction, ip_addr) > { > var re = /^sofia\//g; > var length = name.search (re); > var new_name = name; > > if (length == -1) > return new_name; > > if (direction == "inbound") { > re = /@.*$/g; > > new_name = name.replace (re, "@" + ip_addr); > } > else if (direction == "outbound") { > re = /\/sip:(.*@[^:]*):.*$/g; > > new_name = name.replace (re, "/$1"); > } > > return new_name; > } > > Suggestions for a better approach? Keep in mind that my existing user > population expects (for better or worse) to use *5 to park the call on > their phone so I'm somewhat limited in what I can do. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: > john at feith.com | > | John Wehle | Fax: 1-215-540-5495 > | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From johnd at defyne.org Wed Jun 10 20:04:22 2009 From: johnd at defyne.org (John Dalgliesh) Date: Thu, 11 Jun 2009 13:04:22 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques Message-ID: Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John From jason at voicesession.com Wed Jun 10 20:09:48 2009 From: jason at voicesession.com (lee jason) Date: Thu, 11 Jun 2009 11:09:48 +0800 Subject: [Freeswitch-users] SIP Registration Address Message-ID: <2cbf225c0906102009vf36fe52gf6ebd8d0c704d44b@mail.gmail.com> Dear All, I just have a question, How can I use Freeswitch to blind two IP address for SIP registration at same port(5060 UDP)? Thanks a lot. Jason Lee > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/2c3ede67/attachment.html From mrene_lists at avgs.ca Wed Jun 10 20:12:19 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 10 Jun 2009 23:12:19 -0400 Subject: [Freeswitch-users] SIP Registration Address In-Reply-To: <2cbf225c0906102009vf36fe52gf6ebd8d0c704d44b@mail.gmail.com> References: <2cbf225c0906102009vf36fe52gf6ebd8d0c704d44b@mail.gmail.com> Message-ID: Hi, Create two sip profiles, one per IP. Math On 10-Jun-09, at 11:09 PM, lee jason wrote: > Dear All, > > I just have a question, How can I use Freeswitch to blind two > IP address for SIP registration at same port(5060 UDP)? > > Thanks a lot. > > Jason Lee > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/bac7391e/attachment.html From mgg at giagnocavo.net Wed Jun 10 21:04:26 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 11 Jun 2009 00:04:26 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> How are you handling your FS box crashing? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Dalgliesh Sent: Wednesday, June 10, 2009 9:04 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Live Upgrade Techniques Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Wed Jun 10 21:41:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 11 Jun 2009 00:41:48 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> Message-ID: <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> By reporting it on Jira so it doesn't crash anymore :D On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote: > How are you handling your FS box crashing? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of John Dalgliesh > Sent: Wednesday, June 10, 2009 9:04 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Live Upgrade Techniques > > > Hi, > > I am slowly gaining confidence using FreeSWITCH in production, but > there > is one issue that I'm still wondering about: how are people upgrading > their FreeSWITCH installation binaries without dropping all current > calls? > > So far I have been upgrading in the dead of night, after pausing for 5 > minutes then dropping the stragglers, but this is hardly ideal. > > What I would like to do is to run an upgraded instance of FreeSWITCH > on > the same machine, and have it handle all new call packets, whereas > the old > instance continues to handle the existing call packets, until there > are no > more old calls left. > > I can think of about seven ways to accomplish this, but before I > dive into > the code I thought I'd better ask what everyone else has been doing :) > > (The only standard way I can think of doing this is to have a SIP > proxy > sitting in front of FS the whole time, just to handle these upgrade > windows. It seems like a bit of a waste.) > > So how are you handling your FS software upgrades? > > {P^/ > John > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Wed Jun 10 22:56:52 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 11 Jun 2009 01:56:52 -0400 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A30547D.4090101@gmail.com> References: <4A30547D.4090101@gmail.com> Message-ID: <4A309CA4.10609@freeswitch.org> jcromes at gmail.com wrote: > Thoughts anyone? > You can't please all of the people all of the time -Ray From diego.viola at gmail.com Wed Jun 10 23:17:15 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 11 Jun 2009 02:17:15 -0400 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A30547D.4090101@gmail.com> References: <4A30547D.4090101@gmail.com> Message-ID: <86a32abc0906102317i7a441a82s877762652b7d92f4@mail.gmail.com> I don't have anything against the web site, I like it, but I kinda agree that these banners are too big and a bit negative/unprofessional. Take this for example: http://www.freeswitch.org/ads/ad8.jpg When someone looks at that the first thing they get is a negative feeling and that's not good, doesn't matter what the message is, the negative feeling is there. So yeah, I kinda agree with you. Regards, Diego On Wed, Jun 10, 2009 at 8:49 PM, wrote: > My company is currently investigating a couple of projects that may take > me in the direction of FreeSwitch... In general, our management does > not often consider open source software for projects such as this, but > I've been successful in proving to them recently that open source can > deliver. > > FreeSwitch is a *very* professional and polished product, I can tell - > from the code and from the community. > > Unfortunately, I've been hesitant to send people to your webpage lately > because it went downhill a few weeks ago. Whenever I think about one of > our executives going to your webpage (after my recommendation) and > seeing a picture of people clanking beer glasses, or some idiot tied up > in phone cables, I cringe. I know you're advertising for ClueCon, but > honestly, some of those huge images on your front page really knock your > product down a peg in professionalism. > > Anyway, I'm pretty new to the community and I don't claim to be a web > designer. You have an excellent piece of software, but if I didn't > already know that about FreeSwitch, your webpage would not make a good > first impression. > > Please take that for what it's worth... I wanted to voice my opinion > because if I'm thinking it, others may be as well. > Thoughts anyone? > > J > > (Have I mentioned how awesome your source browser is though??!!) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/74e7bd07/attachment.html From michal.bielicki at halo2.pl Thu Jun 11 03:47:46 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Thu, 11 Jun 2009 12:47:46 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: Message-ID: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> Am 11.06.2009 um 05:04 schrieb John Dalgliesh: > > Hi, > > I am slowly gaining confidence using FreeSWITCH in production, but > there > is one issue that I'm still wondering about: how are people upgrading > their FreeSWITCH installation binaries without dropping all current > calls? > > So far I have been upgrading in the dead of night, after pausing for 5 > minutes then dropping the stragglers, but this is hardly ideal. > > What I would like to do is to run an upgraded instance of FreeSWITCH > on > the same machine, and have it handle all new call packets, whereas > the old > instance continues to handle the existing call packets, until there > are no > more old calls left. > > I can think of about seven ways to accomplish this, but before I > dive into > the code I thought I'd better ask what everyone else has been doing :) > > (The only standard way I can think of doing this is to have a SIP > proxy > sitting in front of FS the whole time, just to handle these upgrade > windows. It seems like a bit of a waste.) > > So how are you handling your FS software upgrades? > > {P^/ > John > > We use freeswitch on solaris and just upgrade it to a new zfs which gets remounted to the old place and freeswitch gracefully restartet. On failure we can allways do a rollback, which takes between 2 and 10 seconds, so the dwntime is pretty acceptable. Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2453 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/93ef242c/attachment.bin From yudha2008 at gmail.com Thu Jun 11 05:13:15 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 11 Jun 2009 17:43:15 +0530 Subject: [Freeswitch-users] FIFO through javascript Message-ID: Hi, I have configured inbound in FS SVN Trunk. i have written small program for inbound call to bridge. i have used session fifo session.execute( "fifo", "sales_fifo_1 out wait undef '/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" ); session.execute("bridge", "sofia/internal/1003%XXXXXXXXXXXX"); Inbound call pass through JavaScript session and play the voice file but it did not bridge to the extension 1003. It keep on playing the same voice file. how can i bridge the call after session.execute. session.execute( "fifo", "sales_fifo_1 out wait undef '/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" ); Can any one assist me to resolve the above problem -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8367970b/attachment.html From brian at freeswitch.org Thu Jun 11 05:21:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 07:21:11 -0500 Subject: [Freeswitch-users] FIFO through javascript In-Reply-To: References: Message-ID: <949646D7-10E9-415F-B908-92AFA5DBF576@freeswitch.org> You can't... once you execute fifo your script has stopped. I think you have the idea that your script will keep running after you enter the fifo... /b On Jun 11, 2009, at 7:13 AM, Baskar wrote: > > Can any one assist me to resolve the above problem From larclap at yahoo.com Thu Jun 11 06:49:09 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 06:49:09 -0700 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log Message-ID: <003901c9ea9b$659eb4a0$30dc1de0$@com> In a dialplan, the action sets effective_caller_id_number to a value, however, in INFO, the displayed value is not the same as the set. Why? http://pastebin.freeswitch.org/9361 Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/e76fd6ca/attachment-0001.html From klaus.teller at gmx.net Thu Jun 11 07:19:47 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 11 Jun 2009 16:19:47 +0200 Subject: [Freeswitch-users] Rejecting calls without answering Message-ID: <20090611141947.246920@gmx.net> Hi Team, I'm still in need of a way to reject a call without answering it. I very much appreciate your help. Klaus. -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 From intralanman at freeswitch.org Thu Jun 11 07:23:22 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 11 Jun 2009 10:23:22 -0400 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <20090611141947.246920@gmx.net> References: <20090611141947.246920@gmx.net> Message-ID: <4A31135A.109@freeswitch.org> Klaus Teller wrote: > Hi Team, > > I'm still in need of a way to reject a call without answering it. I very much appreciate your help. > > Klaus. > -Ray From brian at freeswitch.org Thu Jun 11 07:24:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 09:24:06 -0500 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <003901c9ea9b$659eb4a0$30dc1de0$@com> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> Message-ID: <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> Are you doing an originate or a bridge? /b On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote: > In a dialplan, the action sets effective_caller_id_number to a > value, however, in INFO, the displayed value is not the same as the > set. Why? > > http://pastebin.freeswitch.org/9361 > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8887882c/attachment.html From brian at freeswitch.org Thu Jun 11 07:24:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 09:24:30 -0500 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <20090611141947.246920@gmx.net> References: <20090611141947.246920@gmx.net> Message-ID: <44BA8761-1819-4B1C-A7BD-4178A37BEFCC@freeswitch.org> respond will do exactly that... try just hangup /b On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: > Hi Team, > > I'm still in need of a way to reject a call without answering it. I > very much appreciate your help. > > Klaus. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/29eb17d2/attachment.html From larclap at yahoo.com Thu Jun 11 07:54:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 07:54:45 -0700 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> Message-ID: <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> Bridge From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 7:24 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log Are you doing an originate or a bridge? /b On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote: In a dialplan, the action sets effective_caller_id_number to a value, however, in INFO, the displayed value is not the same as the set. Why? http://pastebin.freeswitch.org/9361 Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/41380844/attachment.html From brian at freeswitch.org Thu Jun 11 08:06:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 10:06:13 -0500 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> Message-ID: <9D918021-7578-4A1F-9C4C-BFF2485B3703@freeswitch.org> make sure you set it before the bridge. /b On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote: > Bridge > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Thursday, June 11, 2009 7:24 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log > > Are you doing an originate or a bridge? > > /b Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/639419f3/attachment-0001.html From anthony.minessale at gmail.com Thu Jun 11 08:17:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 10:17:35 -0500 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A30547D.4090101@gmail.com> References: <4A30547D.4090101@gmail.com> Message-ID: <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> One important thing is that if we go around following everything everybody else says we become a follower in our field. I have had numerous people tell me what to do in the code, what to name things, what to eat for breakfast. Plain and simple, I will choose what to put on our website, when to put it there and what it says. You are welcome to your own opinion. I have no problem with it. If you say something I like we may even listen. Feel free to comment on anything else you find when browsing our community. BUT, If you have no sense of humor, you will not make it far in the open source telecom industry. If you want a more professional looking site, we do have some guys in suits on our FreeSWITCH Solutions site. http://www.freeswitchsolutions.com/ On Wed, Jun 10, 2009 at 7:49 PM, wrote: > My company is currently investigating a couple of projects that may take > me in the direction of FreeSwitch... In general, our management does > not often consider open source software for projects such as this, but > I've been successful in proving to them recently that open source can > deliver. > > FreeSwitch is a *very* professional and polished product, I can tell - > from the code and from the community. > > Unfortunately, I've been hesitant to send people to your webpage lately > because it went downhill a few weeks ago. Whenever I think about one of > our executives going to your webpage (after my recommendation) and > seeing a picture of people clanking beer glasses, or some idiot tied up > in phone cables, I cringe. I know you're advertising for ClueCon, but > honestly, some of those huge images on your front page really knock your > product down a peg in professionalism. > > Anyway, I'm pretty new to the community and I don't claim to be a web > designer. You have an excellent piece of software, but if I didn't > already know that about FreeSwitch, your webpage would not make a good > first impression. > > Please take that for what it's worth... I wanted to voice my opinion > because if I'm thinking it, others may be as well. > Thoughts anyone? > > J > > (Have I mentioned how awesome your source browser is though??!!) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/18121d58/attachment.html From larclap at yahoo.com Thu Jun 11 08:26:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 08:26:45 -0700 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <9D918021-7578-4A1F-9C4C-BFF2485B3703@freeswitch.org> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> <9D918021-7578-4A1F-9C4C-BFF2485B3703@freeswitch.org> Message-ID: <007e01c9eaa9$07e8ce00$17ba6a00$@com> It was. You can see the set at line 2 was done before the bridge at line 4. What am I missing? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 8:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log make sure you set it before the bridge. /b On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote: Bridge From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 7:24 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log Are you doing an originate or a bridge? /b Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/6d428e66/attachment.html From steveu at coppice.org Thu Jun 11 09:09:06 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 12 Jun 2009 00:09:06 +0800 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> Message-ID: <4A312C22.1010504@coppice.org> Anthony Minessale wrote: > One important thing is that if we go around following everything > everybody else says > we become a follower in our field. > > I have had numerous people tell me what to do in the code, what to > name things, what to eat for breakfast. > Plain and simple, I will choose what to put on our website, when to > put it there and what it says. > > You are welcome to your own opinion. I have no problem with it. If > you say something I like we may > even listen. Feel free to comment on anything else you find when > browsing our community. > > BUT, > > If you have no sense of humor, you will not make it far in the open > source telecom industry. > > If you want a more professional looking site, we do have some guys in > suits on our FreeSWITCH Solutions site. > http://www.freeswitchsolutions.com/ The main reason www.freeswitchsolutions.com looks more professional that www.freeswitch.org is not the content of the pictures but their size. The pictures at the top of the www.freeswitch.org are too big and in your face. They completely dominate the screen when it appears. Steve From klaus.teller at gmx.net Thu Jun 11 09:21:30 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 11 Jun 2009 18:21:30 +0200 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <44BA8761-1819-4B1C-A7BD-4178A37BEFCC@freeswitch.org> References: <20090611141947.246920@gmx.net> <44BA8761-1819-4B1C-A7BD-4178A37BEFCC@freeswitch.org> Message-ID: <20090611162130.284160@gmx.net> Hi Folks, Here is what i'm observing. When i connect with Xlite (registered device) and call the 9444 extension (see below), Freeswitch does hangup as i would like it to. But when i call via gafachi, something weird happens. What i can see is that Freeswitch sends a hangup signal (service temporarily not available) to Gafachi, but the guys keep sending back the very same call. It looks to me like a Gafachi issue. But can anything else be done on the Freeswitch side? I'm attaching the logs for the gafachi call this. All you see in there is just one single call. You will see that a new channel is created more than once. Any thought? Klaus. The gafachi respond extension (under conf/dialplan/public/reject.xml): The gafachi profile (under conf/sip_profiles/external/gafachi.xml): The Xlite respond test extension (in default.xml): Any idea? > respond will do exactly that... try just hangup > > /b > > On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: > > > Hi Team, > > > > I'm still in need of a way to reject a call without answering it. I > > very much appreciate your help. > > > > Klaus. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 -------------- next part -------------- 2009-06-11 12:14:34.870820 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [6867d2a1-e26a-4cab-9122-7ecab2b9397f] 2009-06-11 12:14:34.870820 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:34.870820 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 273544867 273544867 IN IP4 67.216.37.18 s=session c=IN IP4 67.216.37.18 t=0 0 m=audio 35116 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:34.870820 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:34.870820 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:34.870820 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_NEW 2009-06-11 12:14:34.874889 [DEBUG] switch_core_state_machine.c:403 (sofia/external/6473671811 at sip.gafachi.com) State NEW 2009-06-11 12:14:34.874889 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:34.874889 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:34.874889 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:34.878827 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:34.878827 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:34.878827 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:34.882811 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:34.882811 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:34.882811 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.882811 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-11 12:14:34.882811 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:34.882811 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:34.882811 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:34.882811 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_CLEARING 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_CLEARING 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_CLEARING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:34.886819 [DEBUG] switch_core_session.c:1067 Session 1 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:34.886819 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:34.886819 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:34.886819 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:38.559058 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [f1544b7b-0176-447d-88f0-db6a422fb89e] 2009-06-11 12:14:38.559058 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:38.559058 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 1686340962 1686340962 IN IP4 67.216.37.18 s=session c=IN IP4 67.216.37.18 t=0 0 m=audio 43928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:38.559058 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:38.559058 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:38.559058 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:38.559058 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:38.559058 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:38.559058 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:38.559058 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:38.559058 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:38.559058 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:38.559058 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:38.563057 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:38.563057 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:38.563057 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:38.563057 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:38.563057 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:38.567118 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:38.567118 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:38.567118 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:1067 Session 2 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:38.567118 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:38.567118 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:38.567118 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:42.471295 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [dcf89ba0-7932-4cc8-a337-5f807ae3c394] 2009-06-11 12:14:42.471295 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:42.471295 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 1254501935 1254501935 IN IP4 67.216.37.17 s=session c=IN IP4 67.216.37.17 t=0 0 m=audio 34272 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:42.471295 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:42.471295 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:42.471295 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:42.471295 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:42.471295 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.471295 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:42.475309 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:42.475309 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:42.475309 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:42.479287 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:42.479287 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:42.479287 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:42.479287 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:42.479287 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:42.479287 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.479287 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:42.479287 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:42.479287 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:42.483297 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:42.483297 [DEBUG] switch_core_session.c:1067 Session 3 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:42.483297 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:42.483297 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:42.483297 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:44.147389 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [7e608276-0901-4525-8822-7d4c07283466] 2009-06-11 12:14:44.147389 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:44.147389 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 36880947 36880947 IN IP4 67.216.37.18 s=session c=IN IP4 67.216.37.18 t=0 0 m=audio 59904 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:44.147389 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:44.147389 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:44.151433 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:44.151433 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:44.151433 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:44.151433 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:44.151433 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:44.155404 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:44.155404 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:44.155404 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:44.155404 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:44.155404 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:1067 Session 4 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:44.155404 [NOTICE] switch_core_session.c:1085 Session 4 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:44.155404 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:46.131565 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [0e736a20-8f6b-4f0a-941a-fe00e277bb80] 2009-06-11 12:14:46.135599 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:46.135599 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 1258094781 1258094781 IN IP4 67.216.37.17 s=session c=IN IP4 67.216.37.17 t=0 0 m=audio 16632 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:46.135599 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:46.135599 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:46.135599 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_NEW 2009-06-11 12:14:46.135599 [DEBUG] switch_core_state_machine.c:403 (sofia/external/6473671811 at sip.gafachi.com) State NEW 2009-06-11 12:14:46.135599 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:46.135599 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:46.135599 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:46.139548 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:46.139548 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:46.139548 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:46.147562 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:46.147562 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:46.147562 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.147562 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:46.147562 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:46.147562 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:46.147562 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:46.147562 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:46.147562 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:46.152178 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:46.152178 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:46.152178 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:46.152178 [DEBUG] switch_core_session.c:1067 Session 5 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:46.152178 [NOTICE] switch_core_session.c:1085 Session 5 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:46.152178 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:46.152178 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:46.157559 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:46.157559 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep From krice at freeswitch.org Thu Jun 11 09:31:13 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Jun 2009 11:31:13 -0500 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <20090611162130.284160@gmx.net> Message-ID: Hah they are just retrying the call to see if they get a different answer the 2nd and 3rd time around... This is common unfortunately since 503 in the VoIP world is typically interpreted by the PSTN world as a "Temp congestion" (and rightly so) so they will retry and not fail the call... You can try responding w/ 486 Busy if you know the call doesn't need to fail somewhere else... > From: Klaus Teller > Reply-To: > Date: Thu, 11 Jun 2009 18:21:30 +0200 > To: > Subject: Re: [Freeswitch-users] Rejecting calls without answering > > Hi Folks, > > Here is what i'm observing. When i connect with Xlite (registered device) and > call the 9444 extension (see below), Freeswitch does hangup as i would like it > to. > > But when i call via gafachi, something weird happens. What i can see is that > Freeswitch sends a hangup signal (service temporarily not available) to > Gafachi, but the guys keep sending back the very same call. > > > It looks to me like a Gafachi issue. But can anything else be done on the > Freeswitch side? > > I'm attaching the logs for the gafachi call this. All you see in there is just > one single call. You will see that a new channel is created more than once. > > Any thought? > > Klaus. > > The gafachi respond extension (under conf/dialplan/public/reject.xml): > > > > > > > > > > The gafachi profile (under conf/sip_profiles/external/gafachi.xml): > > > > > > > > > The Xlite respond test extension (in default.xml): > > > > > > > > Any idea? > > > > > > >> respond will do exactly that... try just hangup >> >> /b >> >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: >> >>> Hi Team, >>> >>> I'm still in need of a way to reject a call without answering it. I >>> very much appreciate your help. >>> >>> Klaus. >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> > > -- > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss > f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jun 11 09:36:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 11:36:47 -0500 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A312C22.1010504@coppice.org> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> <4A312C22.1010504@coppice.org> Message-ID: <191c3a030906110936i1ad40654y4635c5532d46150@mail.gmail.com> Ok, let's try them at half size. On Thu, Jun 11, 2009 at 11:09 AM, Steve Underwood wrote: > Anthony Minessale wrote: > > One important thing is that if we go around following everything > > everybody else says > > we become a follower in our field. > > > > I have had numerous people tell me what to do in the code, what to > > name things, what to eat for breakfast. > > Plain and simple, I will choose what to put on our website, when to > > put it there and what it says. > > > > You are welcome to your own opinion. I have no problem with it. If > > you say something I like we may > > even listen. Feel free to comment on anything else you find when > > browsing our community. > > > > BUT, > > > > If you have no sense of humor, you will not make it far in the open > > source telecom industry. > > > > If you want a more professional looking site, we do have some guys in > > suits on our FreeSWITCH Solutions site. > > http://www.freeswitchsolutions.com/ > The main reason www.freeswitchsolutions.com > looks more professional that > www.freeswitch.org is not the content of the pictures but their size. > The pictures at the top of the www.freeswitch.org are too big and in > your face. They completely dominate the screen when it appears. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/2f8168d0/attachment.html From klaus.teller at gmx.net Thu Jun 11 09:40:24 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 11 Jun 2009 18:40:24 +0200 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: References: Message-ID: <20090611164024.22470@gmx.net> Excellent! Thank you everybody. Response 486 did the trick. Klaus. -------- Original-Nachricht -------- > Datum: Thu, 11 Jun 2009 11:31:13 -0500 > Von: Ken Rice > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Rejecting calls without answering > Hah they are just retrying the call to see if they get a different answer > the 2nd and 3rd time around... This is common unfortunately since 503 in > the > VoIP world is typically interpreted by the PSTN world as a "Temp > congestion" > (and rightly so) so they will retry and not fail the call... You can try > responding w/ 486 Busy if you know the call doesn't need to fail somewhere > else... > > > > From: Klaus Teller > > Reply-To: > > Date: Thu, 11 Jun 2009 18:21:30 +0200 > > To: > > Subject: Re: [Freeswitch-users] Rejecting calls without answering > > > > Hi Folks, > > > > Here is what i'm observing. When i connect with Xlite (registered > device) and > > call the 9444 extension (see below), Freeswitch does hangup as i would > like it > > to. > > > > But when i call via gafachi, something weird happens. What i can see is > that > > Freeswitch sends a hangup signal (service temporarily not available) to > > Gafachi, but the guys keep sending back the very same call. > > > > > > It looks to me like a Gafachi issue. But can anything else be done on > the > > Freeswitch side? > > > > I'm attaching the logs for the gafachi call this. All you see in there > is just > > one single call. You will see that a new channel is created more than > once. > > > > Any thought? > > > > Klaus. > > > > The gafachi respond extension (under conf/dialplan/public/reject.xml): > > > > > > expression="^866.*$"> > > > > > > > > > > > > > > The gafachi profile (under conf/sip_profiles/external/gafachi.xml): > > > > > > > > > > > > > > > > > > The Xlite respond test extension (in default.xml): > > > > > > > > > > > > > > > > Any idea? > > > > > > > > > > > > > >> respond will do exactly that... try just hangup > >> > >> /b > >> > >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: > >> > >>> Hi Team, > >>> > >>> I'm still in need of a way to reject a call without answering it. I > >>> very much appreciate your help. > >>> > >>> Klaus. > >> > >> Brian West > >> brian at freeswitch.org > >> > >> -- Meet us at ClueCon! http://www.cluecon.com > >> > >> > >> > >> > > > > -- > > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und > Telefonanschluss > > f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From diego.viola at gmail.com Thu Jun 11 09:45:56 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 11 Jun 2009 12:45:56 -0400 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A312C22.1010504@coppice.org> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> <4A312C22.1010504@coppice.org> Message-ID: <86a32abc0906110945w27829e4ei1cff1cbfe3409bac@mail.gmail.com> That's what I tried to say, I didn't expressed myself well, sorry. On Thu, Jun 11, 2009 at 12:09 PM, Steve Underwood wrote: > Anthony Minessale wrote: > > One important thing is that if we go around following everything > > everybody else says > > we become a follower in our field. > > > > I have had numerous people tell me what to do in the code, what to > > name things, what to eat for breakfast. > > Plain and simple, I will choose what to put on our website, when to > > put it there and what it says. > > > > You are welcome to your own opinion. I have no problem with it. If > > you say something I like we may > > even listen. Feel free to comment on anything else you find when > > browsing our community. > > > > BUT, > > > > If you have no sense of humor, you will not make it far in the open > > source telecom industry. > > > > If you want a more professional looking site, we do have some guys in > > suits on our FreeSWITCH Solutions site. > > http://www.freeswitchsolutions.com/ > The main reason www.freeswitchsolutions.com > looks more professional that > www.freeswitch.org is not the content of the pictures but their size. > The pictures at the top of the www.freeswitch.org are too big and in > your face. They completely dominate the screen when it appears. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/fc025a16/attachment.html From freeswitch-users-list at metik.com Thu Jun 11 09:53:28 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 11 Jun 2009 12:53:28 -0400 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: References: Message-ID: Although a little overkill, RFC3398 also describes some desirable interop behavior between ISUP, ISDN and SIP. (From "7.2.4.1 ISDN Cause Code to Status Code Mapping") [...] ISUP Cause value SIP response ---------------- ------------ 1 unallocated number 404 Not Found 2 no route to network 404 Not found 3 no route to destination 404 Not found 16 normal call clearing --- (*) 17 user busy 486 Busy here 18 no user responding 408 Request Timeout 19 no answer from the user 480 Temporarily unavailable 20 subscriber absent 480 Temporarily unavailable 21 call rejected 403 Forbidden (+) 22 number changed (w/o diagnostic) 410 Gone 22 number changed (w/ diagnostic) 301 Moved Permanently 23 redirection to new destination 410 Gone 26 non-selected user clearing 404 Not Found (=) 27 destination out of order 502 Bad Gateway 28 address incomplete 484 Address incomplete 29 facility rejected 501 Not implemented 31 normal unspecified 480 Temporarily unavailable [...] Resource unavailable This kind of cause value indicates a temporary failure. A 'Retry-After' header MAY be added to the response if appropriate. ISUP Cause value SIP response ---------------- ------------ 34 no circuit available 503 Service unavailable 38 network out of order 503 Service unavailable 41 temporary failure 503 Service unavailable 42 switching equipment congestion 503 Service unavailable 47 resource unavailable 503 Service unavailable ----- Original Message ----- From: "Ken Rice" To: Sent: Thursday, June 11, 2009 12:31 PM Subject: Re: [Freeswitch-users] Rejecting calls without answering Hah they are just retrying the call to see if they get a different answer the 2nd and 3rd time around... This is common unfortunately since 503 in the VoIP world is typically interpreted by the PSTN world as a "Temp congestion" (and rightly so) so they will retry and not fail the call... You can try responding w/ 486 Busy if you know the call doesn't need to fail somewhere else... > From: Klaus Teller > Reply-To: > Date: Thu, 11 Jun 2009 18:21:30 +0200 > To: > Subject: Re: [Freeswitch-users] Rejecting calls without answering > > Hi Folks, > > Here is what i'm observing. When i connect with Xlite (registered device) > and > call the 9444 extension (see below), Freeswitch does hangup as i would > like it > to. > > But when i call via gafachi, something weird happens. What i can see is > that > Freeswitch sends a hangup signal (service temporarily not available) to > Gafachi, but the guys keep sending back the very same call. > > > It looks to me like a Gafachi issue. But can anything else be done on the > Freeswitch side? > > I'm attaching the logs for the gafachi call this. All you see in there is > just > one single call. You will see that a new channel is created more than > once. > > Any thought? > > Klaus. > > The gafachi respond extension (under conf/dialplan/public/reject.xml): > > > > > > > > > > The gafachi profile (under conf/sip_profiles/external/gafachi.xml): > > > > > > > > > The Xlite respond test extension (in default.xml): > > > > > > > > Any idea? > > > > > > >> respond will do exactly that... try just hangup >> >> /b >> >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: >> >>> Hi Team, >>> >>> I'm still in need of a way to reject a call without answering it. I >>> very much appreciate your help. >>> >>> Klaus. >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> > > -- > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss > f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jun 11 09:54:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 11:54:32 -0500 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> Message-ID: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> or you can put a sip proxy in front of 2 boxes where you can control the flow of traffic. when you want to upgrade one, take all the traffic off of it by forcing all calls to the other box, upgrade it then shift the traffic to the new one. if that goes well, upgrade the other one too. On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki wrote: > > Am 11.06.2009 um 05:04 schrieb John Dalgliesh: > > >> Hi, >> >> I am slowly gaining confidence using FreeSWITCH in production, but there >> is one issue that I'm still wondering about: how are people upgrading >> their FreeSWITCH installation binaries without dropping all current calls? >> >> So far I have been upgrading in the dead of night, after pausing for 5 >> minutes then dropping the stragglers, but this is hardly ideal. >> >> What I would like to do is to run an upgraded instance of FreeSWITCH on >> the same machine, and have it handle all new call packets, whereas the old >> instance continues to handle the existing call packets, until there are no >> more old calls left. >> >> I can think of about seven ways to accomplish this, but before I dive into >> the code I thought I'd better ask what everyone else has been doing :) >> >> (The only standard way I can think of doing this is to have a SIP proxy >> sitting in front of FS the whole time, just to handle these upgrade >> windows. It seems like a bit of a waste.) >> >> So how are you handling your FS software upgrades? >> >> {P^/ >> John >> >> >> > > We use freeswitch on solaris and just upgrade it to a new zfs which gets > remounted to the old place and freeswitch gracefully restartet. On failure > we can allways do a rollback, which takes between 2 and 10 seconds, so the > dwntime is pretty acceptable. > > Michal Bielicki > Leiter der Niederlassung > HaloKwadrat Sp. z o.o. > Niederlassung Kleinmachnow > Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P > Ust.Id.: DE261885536 > Geschaeftsfuehrer: Aleksander Wiercinski > Meiereifeld 2b, 14532 Kleinmachnow > t. +49 33203 263220 > f. +49 33203 263229 sip. info at halokwadrat.de > e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de > Hauptgesch?ftsstelle: > Halo Kwadrat Sp. z o.o. > ul. Polna 46/14 > 00-644 Warszawa, Polen > EIngetragen im HRB Warszawa, KRS 0000153539 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/dda20ded/attachment.html From mgg at giagnocavo.net Thu Jun 11 10:33:53 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 11 Jun 2009 13:33:53 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> Exactly. You probably want to have something like this anyways, so that when someone accidentally unplugs the system, or the disks/CPU/RAM crash, you're not stuck. That is, until FreeSWITCH can record its internal state to some inter-machine memory so we can have hot failover. ;) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, June 11, 2009 10:55 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques or you can put a sip proxy in front of 2 boxes where you can control the flow of traffic. when you want to upgrade one, take all the traffic off of it by forcing all calls to the other box, upgrade it then shift the traffic to the new one. if that goes well, upgrade the other one too. On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki wrote: Am 11.06.2009 um 05:04 schrieb John Dalgliesh: Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John We use freeswitch on solaris and just upgrade it to a new zfs which gets remounted to the old place and freeswitch gracefully restartet. On failure we can allways do a rollback, which takes between 2 and 10 seconds, so the dwntime is pretty acceptable. Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c0762d9d/attachment.html From msc at freeswitch.org Thu Jun 11 10:42:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 10:42:49 -0700 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> Message-ID: <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote: > Exactly. You probably want to have something like this anyways, so that > when someone accidentally unplugs the system, or the disks/CPU/RAM crash, > you?re not stuck. > > > > That is, until FreeSWITCH can record its internal state to some > inter-machine memory so we can have hot failover. ;) > > > I think that's going to be in 1.0.5. :) > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Thursday, June 11, 2009 10:55 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques > > > > or you can put a sip proxy in front of 2 boxes where you can control the > flow of traffic. > when you want to upgrade one, take all the traffic off of it by forcing all > calls to the other box, upgrade it then shift the traffic to the new one. > if that goes well, upgrade the other one too. > > > On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki > wrote: > > > Am 11.06.2009 um 05:04 schrieb John Dalgliesh: > > > > > Hi, > > I am slowly gaining confidence using FreeSWITCH in production, but there > is one issue that I'm still wondering about: how are people upgrading > their FreeSWITCH installation binaries without dropping all current calls? > > So far I have been upgrading in the dead of night, after pausing for 5 > minutes then dropping the stragglers, but this is hardly ideal. > > What I would like to do is to run an upgraded instance of FreeSWITCH on > the same machine, and have it handle all new call packets, whereas the old > instance continues to handle the existing call packets, until there are no > more old calls left. > > I can think of about seven ways to accomplish this, but before I dive into > the code I thought I'd better ask what everyone else has been doing :) > > (The only standard way I can think of doing this is to have a SIP proxy > sitting in front of FS the whole time, just to handle these upgrade > windows. It seems like a bit of a waste.) > > So how are you handling your FS software upgrades? > > {P^/ > John > > > > We use freeswitch on solaris and just upgrade it to a new zfs which gets > remounted to the old place and freeswitch gracefully restartet. On failure > we can allways do a rollback, which takes between 2 and 10 seconds, so the > dwntime is pretty acceptable. > > Michal Bielicki > Leiter der Niederlassung > HaloKwadrat Sp. z o.o. > Niederlassung Kleinmachnow > Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P > Ust.Id.: DE261885536 > Geschaeftsfuehrer: Aleksander Wiercinski > Meiereifeld 2b, 14532 Kleinmachnow > t. +49 33203 263220 > f. +49 33203 263229 sip. info at halokwadrat.de > e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de > Hauptgesch?ftsstelle: > Halo Kwadrat Sp. z o.o. > ul. Polna 46/14 > 00-644 Warszawa, Polen > EIngetragen im HRB Warszawa, KRS 0000153539 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/030512c7/attachment-0001.html From jcromes at gmail.com Thu Jun 11 10:43:40 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Thu, 11 Jun 2009 12:43:40 -0500 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> Message-ID: <4A31424C.4020501@gmail.com> Haha, good point on the FreeSwitch Solutions site... Suits - very professional. =) Please don't think I'm telling you guys what to do, I know you don't need that. It IS your software and your site, and you've done a HELL of a job with it so far. It was just a thought. Sorry if I offended. Anthony Minessale wrote: > One important thing is that if we go around following everything > everybody else says > we become a follower in our field. > > I have had numerous people tell me what to do in the code, what to > name things, what to eat for breakfast. > Plain and simple, I will choose what to put on our website, when to > put it there and what it says. > > You are welcome to your own opinion. I have no problem with it. If > you say something I like we may > even listen. Feel free to comment on anything else you find when > browsing our community. > > BUT, > > If you have no sense of humor, you will not make it far in the open > source telecom industry. > > If you want a more professional looking site, we do have some guys in > suits on our FreeSWITCH Solutions site. > http://www.freeswitchsolutions.com/ > From msc at freeswitch.org Thu Jun 11 10:47:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 10:47:24 -0700 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A31424C.4020501@gmail.com> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> <4A31424C.4020501@gmail.com> Message-ID: <87f2f3b90906111047v1febdbffsdc0b59528980b1e8@mail.gmail.com> On Thu, Jun 11, 2009 at 10:43 AM, wrote: > Haha, good point on the FreeSwitch Solutions site... Suits - very > professional. =) > Please don't think I'm telling you guys what to do, I know you don't > need that. It IS your software and your site, and you've done a HELL of > a job with it so far. > > It was just a thought. Sorry if I offended. > No offense taken. We DO appreciate feedback, unsolicited or otherwise, but we don't always agree with it. Definitely show your bosses the FSS site. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/500b8c81/attachment.html From johnd at defyne.org Thu Jun 11 11:00:24 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 04:00:24 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> Message-ID: OK thanks that is what I thought the general way of doing it would be. But it seems a bit wasteful to have that SIP proxy there the whole time especially when I am using FS in the role of an SBC. The problem with the graceful restart of course is that you have to wait for the calls count to get to zero, which may never happen. It's 3:30am here in Sydney now and I just checked FS: 20 calls in progress still! So what I plan to do is add a '--upgrade' cmd line arg to FS. This will make the new instance contact the old one on a unix socket and receive a dup of its SIP socket fd(s) via a SCM_RIGHTS sendmsg. It will use those for sending and the unix socket for receiving. Meanwhile the old instance will pass any packets with unknown Call-Ids over the unix socket to the new instance, instead of handling them itself. When the old instance has no calls left, it shuts down. The new instance detects the unix socket is closed and switches to reading from the SIP socket (which would have buffered any unread packets - so nothing is lost). Sound good? I realise this will be 90% in libsofia but I've read teh code and it seems very do-able. Anyone interested in my changes will of course be most welcome to them. The runner-up approach I considered was to make a kernel module that extends iptables with a filter that can extract the Call-Id and look it up in a table that is somehow populated from FS. Maybe this exists already? Kind of a SIP proxy lite that can be enabled on the server machine when needed. Anyway that lost out as it's more work and even less portable. {P^/ John On Thu, 11 Jun 2009 at 11:54 -0500, Anthony Minessale wrote: > > or you can put a sip proxy in front of 2 boxes where you can control the > flow of traffic. > when you want to upgrade one, take all the traffic off of it by forcing all > calls to the other box, upgrade it then shift the traffic to the new one. > if that goes well, upgrade the other one too. > > > > On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki > wrote: > >> >> Am 11.06.2009 um 05:04 schrieb John Dalgliesh: >> >> >>> Hi, >>> >>> I am slowly gaining confidence using FreeSWITCH in production, but there >>> is one issue that I'm still wondering about: how are people upgrading >>> their FreeSWITCH installation binaries without dropping all current calls? >>> >>> So far I have been upgrading in the dead of night, after pausing for 5 >>> minutes then dropping the stragglers, but this is hardly ideal. >>> >>> What I would like to do is to run an upgraded instance of FreeSWITCH on >>> the same machine, and have it handle all new call packets, whereas the old >>> instance continues to handle the existing call packets, until there are no >>> more old calls left. >>> >>> I can think of about seven ways to accomplish this, but before I dive into >>> the code I thought I'd better ask what everyone else has been doing :) >>> >>> (The only standard way I can think of doing this is to have a SIP proxy >>> sitting in front of FS the whole time, just to handle these upgrade >>> windows. It seems like a bit of a waste.) >>> >>> So how are you handling your FS software upgrades? >>> >>> {P^/ >>> John >>> >>> >>> >> >> We use freeswitch on solaris and just upgrade it to a new zfs which gets >> remounted to the old place and freeswitch gracefully restartet. On failure >> we can allways do a rollback, which takes between 2 and 10 seconds, so the >> dwntime is pretty acceptable. >> >> Michal Bielicki >> Leiter der Niederlassung >> HaloKwadrat Sp. z o.o. >> Niederlassung Kleinmachnow >> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P >> Ust.Id.: DE261885536 >> Geschaeftsfuehrer: Aleksander Wiercinski >> Meiereifeld 2b, 14532 Kleinmachnow >> t. +49 33203 263220 >> f. +49 33203 263229 sip. info at halokwadrat.de >> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de >> Hauptgesch?ftsstelle: >> Halo Kwadrat Sp. z o.o. >> ul. Polna 46/14 >> 00-644 Warszawa, Polen >> EIngetragen im HRB Warszawa, KRS 0000153539 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > From lcm at marshap.com Thu Jun 11 11:04:07 2009 From: lcm at marshap.com (Larry Marshall) Date: Thu, 11 Jun 2009 11:04:07 -0700 Subject: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition Message-ID: <00d501c9eabf$03c7ab00$0b570100$@com> http://pastebin.freeswitch.org/9365 I do not know what I am doing wrong. I am trying to set the effective_caller_id_name and _number depending on the originating extension. I tried: and and But each got substituted with the name of the extension in the log: Dialplan: sofia/internal/1000 at 192.168.10.29 Regex (FAIL) [Long Distance - flowroute] () =~ /^100[09]$/ break=on-true where the extension looks like: Info from the log shows variable_sip_from_user: [1000] Caller-Caller-ID-Number: [1000] Can anyone help? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8be3a0b6/attachment.html From johnd at defyne.org Thu Jun 11 11:13:39 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 04:13:39 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> Message-ID: I assume he's talking about hardware failures here :P But to answer the question: crashes are easy to deal with. With a crash you have lost the calls that are in progress anyway; you don't have to manage a gradual transition. Currently, since FS is quite quick to start up, I am just relaunching it immediately. But when I have a second box up and running what I'll do is just add the IP of the dead machine as another IP of the second box, and then it will take all the old machine's traffic. That is the plan anyway. I've seen some commercial boxes that use a similar trick. (I've only seen one crash that wasn't my fault. Something to do with terminating a bridge: when the first leg gets a hangup it hangs up the other leg on its own thread... which can cause problems if the other leg was doing something funky at the time. Leads to a heap corruption. Doesn't happen with MALLOC_CHECK_ set so I'm just leaving it set for now :) {P^/ On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote: > > By reporting it on Jira so it doesn't crash anymore :D > > > On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote: > >> How are you handling your FS box crashing? >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of John Dalgliesh >> Sent: Wednesday, June 10, 2009 9:04 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Live Upgrade Techniques >> >> >> Hi, >> >> I am slowly gaining confidence using FreeSWITCH in production, but >> there >> is one issue that I'm still wondering about: how are people upgrading >> their FreeSWITCH installation binaries without dropping all current >> calls? >> >> So far I have been upgrading in the dead of night, after pausing for 5 >> minutes then dropping the stragglers, but this is hardly ideal. >> >> What I would like to do is to run an upgraded instance of FreeSWITCH >> on >> the same machine, and have it handle all new call packets, whereas >> the old >> instance continues to handle the existing call packets, until there >> are no >> more old calls left. >> >> I can think of about seven ways to accomplish this, but before I >> dive into >> the code I thought I'd better ask what everyone else has been doing :) >> >> (The only standard way I can think of doing this is to have a SIP >> proxy >> sitting in front of FS the whole time, just to handle these upgrade >> windows. It seems like a bit of a waste.) >> >> So how are you handling your FS software upgrades? >> >> {P^/ >> John >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From johnd at defyne.org Thu Jun 11 11:24:10 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 04:24:10 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> Message-ID: On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote: > On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote: >> >> Exactly. You probably want to have something like this anyways, so that >> when someone accidentally unplugs the system, or the disks/CPU/RAM crash, >> you?re not stuck. >> >> >> >> That is, until FreeSWITCH can record its internal state to some >> inter-machine memory so we can have hot failover. ;) >> >> >> > I think that's going to be in 1.0.5. :) I'm still too much of a noob to be certain that's a joke :) ... but FS core already does record much of its internal state... to a DB, right? It just has to not clear that out on startup and problem solved! OTOH there will be a bit of trouble getting the internal state out of all those modules and libraries... in particular sofia :D {P^/ From larclap at yahoo.com Thu Jun 11 11:27:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 11:27:59 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <87f2f3b90906101718g72911b8fv326ee9c470fdf097@mail.gmail.com> References: <005001c9e8a9$76c5c780$64515680$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> <015b01c9ea26$d0abb2e0$720318a0$@com> <87f2f3b90906101718g72911b8fv326ee9c470fdf097@mail.gmail.com> Message-ID: <011001c9eac2$59884e70$0c98eb50$@com> Michael, Removing everything between the tag in sip_profiles/internal/example.xml did the trick - no error message on FS startup. I'm running 13723. 2009-06-11 07:21:03.609317 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-11 07:21:03.612274 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-11 07:21:03.995025 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-11 07:21:03.995436 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-11 07:21:04.245056 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-11 07:21:04.246056 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-11 07:21:04.745950 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-11 07:21:05.745725 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-11 07:21:07.745256 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-11 07:21:12.221251 [DEBUG] switch_nat.c:77 No InternetGatewayDevice, using first entry as default. 2009-06-11 07:21:12.234867 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 10, 2009 5:18 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb wrote: Rupa, I think the console log has information in it that log/freeswitch.log does not. Console: [root at fs bin]# ../freeswitch 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (Invalid or incomplete multibyte or wide character) 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! log/freeswitch.log: 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock interface 'console' to wait for existing references. (from previous Freeswitch invocation) 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding Dialplan 'enum' 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding Application 'enum' 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum' 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum_auto' 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default template. 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template example. 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template linksys. 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template asterisk. I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console log. However, the disk log begins at 16:12:58, whereas the console log starts at 16:12:50. The console log finishes its NAT and UPnP reporting before the disk log begins, so I wouldn't see any 0.0.0.0 if it were present. The [ERR] was due to me removing example.xml from sip_profiles/internal. I put it back after this. I don't understand the following command in conf/sofia.conf.xml. I think this is just a cosmetic error. You could probably put an empty xml file in sip_profiles/internal and be done with it. Or possibly have just an empty include node, like "" Try it out and report back - we're dying to know what happens! ;) -MC Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 6:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/b45b3d0c/attachment-0001.html From msc at freeswitch.org Thu Jun 11 11:29:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 11:29:15 -0700 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> Message-ID: <87f2f3b90906111129w5f5c5678lf05b025f57447e06@mail.gmail.com> On Thu, Jun 11, 2009 at 11:24 AM, John Dalgliesh wrote: > On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote: > >> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote: >> >>> >>> Exactly. You probably want to have something like this anyways, so that >>> when someone accidentally unplugs the system, or the disks/CPU/RAM crash, >>> you?re not stuck. >>> >>> >>> >>> That is, until FreeSWITCH can record its internal state to some >>> inter-machine memory so we can have hot failover. ;) >>> >>> >>> >>> I think that's going to be in 1.0.5. :) >> > > I'm still too much of a noob to be certain that's a joke :) ... but FS core > already does record much of its internal state... to a DB, right? It just > has to not clear that out on startup and problem solved! > It was a joke. :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/e9bf3fbf/attachment.html From mike at jerris.com Thu Jun 11 11:30:57 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Jun 2009 14:30:57 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> Message-ID: <74170625-2AA0-4FF6-969D-3C850DCD7CA0@jerris.com> On Jun 11, 2009, at 2:24 PM, John Dalgliesh wrote: > On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote: >> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote: >>> >>> Exactly. You probably want to have something like this anyways, so >>> that >>> when someone accidentally unplugs the system, or the disks/CPU/RAM >>> crash, >>> you?re not stuck. >>> >>> That is, until FreeSWITCH can record its internal state to some >>> inter-machine memory so we can have hot failover. ;) >>> >> I think that's going to be in 1.0.5. :) > > I'm still too much of a noob to be certain that's a joke :) ... but > FS core already does record much of its internal state... to a DB, > right? It just has to not clear that out on startup and problem > solved! > > OTOH there will be a bit of trouble getting the internal state out > of all those modules and libraries... in particular sofia :D We have talked quite some about this, its a major job, easily months of work for multiple programmers. We would love to do it but its not on any roadmaps at this time. Mike From brian at freeswitch.org Thu Jun 11 11:07:14 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 13:07:14 -0500 Subject: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition In-Reply-To: <00d501c9eabf$03c7ab00$0b570100$@com> References: <00d501c9eabf$03c7ab00$0b570100$@com> Message-ID: <6603D022-5AD5-4416-9978-8F0E20AAD3A8@freeswitch.org> try destination_number /b On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote: > http://pastebin.freeswitch.org/9365 > > I do not know what I am doing wrong. I am trying to set the > effective_caller_id_name and _number depending on the originating > extension. > > I tried: > expression="^100[09]$" break="on-true"> > and > expression="^100[09]$" break="on-true"> > and > expression="^100[09]$" break="on-true"> > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/7c568749/attachment.html From mrene_lists at avgs.ca Thu Jun 11 11:38:10 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 11 Jun 2009 14:38:10 -0400 Subject: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition In-Reply-To: <6603D022-5AD5-4416-9978-8F0E20AAD3A8@freeswitch.org> References: <00d501c9eabf$03c7ab00$0b570100$@com> <6603D022-5AD5-4416-9978-8F0E20AAD3A8@freeswitch.org> Message-ID: Your syntax is also wrong, it should be and NOT field=${varname}$ Math On 11-Jun-09, at 2:07 PM, Brian West wrote: > try destination_number > > /b > > On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote: > >> http://pastebin.freeswitch.org/9365 >> >> I do not know what I am doing wrong. I am trying to set the >> effective_caller_id_name and _number depending on the originating >> extension. >> >> I tried: >> > expression="^100[09]$" break="on-true"> >> and >> > expression="^100[09]$" break="on-true"> >> and >> > expression="^100[09]$" break="on-true"> >> > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8ca17349/attachment.html From kristian.kielhofner at gmail.com Thu Jun 11 11:41:51 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 11 Jun 2009 14:41:51 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> Message-ID: <2d9149cd0906111141m66a7fe1ao6bc29f520fb08f06@mail.gmail.com> That's exactly what I do. Between dispatcher and FLAGS/GFLAGS this is easy to do in OpenSIPS/SER. On Thu, Jun 11, 2009 at 12:54 PM, Anthony Minessale wrote: > or you can put a sip proxy in front of 2 boxes where you can control the > flow of traffic. > when you want to upgrade one, take all the traffic off of it by forcing all > calls to the other box, upgrade it then shift the traffic to the new one. > if that goes well, upgrade the other one too. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From larclap at yahoo.com Thu Jun 11 12:49:24 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 12:49:24 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match Message-ID: <016601c9eacd$b8ee4cb0$2acae610$@com> I have a match expression for outbound calls as "\d{10}". It's fine for unformatted numbers. Not knowing any better, I created another extension to handle numbers formatted like XXX-XXX-XXXX, which is easier to read and exists in one hard phone's phonebook. It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many extensions for different formats. There's got to be a better way. Any suggestions? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/ebb49988/attachment.html From brian at freeswitch.org Thu Jun 11 12:57:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 14:57:52 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <016601c9eacd$b8ee4cb0$2acae610$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> Message-ID: <3ED39806-9163-4593-B1AE-E4655EAF625B@freeswitch.org> Don't do it! Doing that stuff is highly silly. /b On Jun 11, 2009, at 2:49 PM, Lars Zeb wrote: > There?s got to be a better way. Any suggestions? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/a57d347b/attachment.html From enno.egbert at web.de Thu Jun 11 13:04:40 2009 From: enno.egbert at web.de (NOx-WHV) Date: Thu, 11 Jun 2009 13:04:40 -0700 (PDT) Subject: [Freeswitch-users] re direct calls Message-ID: <23982889.post@talk.nabble.com> Hello Freeswitch User! I am using FS since a few weeks. My intent is to have clients who uses TLS and SRTP for a full encrypted call. I just managed it, that calls are encrypted with TLS and SRTP. My second aim ist to redirect this calls for reduce the processing of the server. I only use the FS for calls between users of this freeswitch. I just tested a "redirect" in the dialplan (without TLS and SRTP) but it doesn?t work. How i have to configure the dialplan for redirect the call to the other user. I use a SNOM hardphone and a phonerlite softphone. Thanks for sour help! NOX -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23982889.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Jun 11 13:11:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 15:11:09 -0500 Subject: [Freeswitch-users] re direct calls In-Reply-To: <23982889.post@talk.nabble.com> References: <23982889.post@talk.nabble.com> Message-ID: On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote: > > Hello Freeswitch User! > > I am using FS since a few weeks. My intent is to have clients who > uses TLS > and SRTP for a full encrypted call. > > I just managed it, that calls are encrypted with TLS and SRTP. My > second aim > ist to redirect this calls for reduce the processing of the server. > I only > use the FS for calls between users of this freeswitch. I just tested a > "redirect" in the dialplan (without TLS and SRTP) but it doesn?t > work. How i > have to configure the dialplan for redirect the call to the other > user. You can't. Its not possible because we are a b2bua and you have already negotiated the keys between the endpoints and FreeSWITCH and when you redirect the media neither phone can decrypt the packets correctly. > > I use a SNOM hardphone and a phonerlite softphone. > > Thanks for sour help! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/bcc62b66/attachment.html From mgg at giagnocavo.net Thu Jun 11 13:33:04 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 11 Jun 2009 16:33:04 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Well, if you're running multiple machines, waiting for it to drainstop isn't that big of a deal unless you're in some sort of hurry, right? Give it an hour or so to drainstop, then kill 'em. Would it not be simpler to try to do something with re-invites or REFER, assuming your endpoints support it? -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Dalgliesh Sent: Thursday, June 11, 2009 12:14 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques I assume he's talking about hardware failures here :P But to answer the question: crashes are easy to deal with. With a crash you have lost the calls that are in progress anyway; you don't have to manage a gradual transition. Currently, since FS is quite quick to start up, I am just relaunching it immediately. But when I have a second box up and running what I'll do is just add the IP of the dead machine as another IP of the second box, and then it will take all the old machine's traffic. That is the plan anyway. I've seen some commercial boxes that use a similar trick. (I've only seen one crash that wasn't my fault. Something to do with terminating a bridge: when the first leg gets a hangup it hangs up the other leg on its own thread... which can cause problems if the other leg was doing something funky at the time. Leads to a heap corruption. Doesn't happen with MALLOC_CHECK_ set so I'm just leaving it set for now :) {P^/ On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote: > > By reporting it on Jira so it doesn't crash anymore :D > > > On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote: > >> How are you handling your FS box crashing? >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of John Dalgliesh >> Sent: Wednesday, June 10, 2009 9:04 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Live Upgrade Techniques >> >> >> Hi, >> >> I am slowly gaining confidence using FreeSWITCH in production, but >> there >> is one issue that I'm still wondering about: how are people upgrading >> their FreeSWITCH installation binaries without dropping all current >> calls? >> >> So far I have been upgrading in the dead of night, after pausing for 5 >> minutes then dropping the stragglers, but this is hardly ideal. >> >> What I would like to do is to run an upgraded instance of FreeSWITCH >> on >> the same machine, and have it handle all new call packets, whereas >> the old >> instance continues to handle the existing call packets, until there >> are no >> more old calls left. >> >> I can think of about seven ways to accomplish this, but before I >> dive into >> the code I thought I'd better ask what everyone else has been doing :) >> >> (The only standard way I can think of doing this is to have a SIP >> proxy >> sitting in front of FS the whole time, just to handle these upgrade >> windows. It seems like a bit of a waste.) >> >> So how are you handling your FS software upgrades? >> >> {P^/ >> John >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:07:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:07:59 +0100 Subject: [Freeswitch-users] Orphaned calls Message-ID: Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,, ,,,,, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/59562fa1/attachment-0001.html From enno.egbert at web.de Thu Jun 11 14:16:07 2009 From: enno.egbert at web.de (NOx-WHV) Date: Thu, 11 Jun 2009 14:16:07 -0700 (PDT) Subject: [Freeswitch-users] re direct calls In-Reply-To: References: <23982889.post@talk.nabble.com> Message-ID: <23989162.post@talk.nabble.com> Thanks for your answer. Can you just announce b2bua. Brian West-3 wrote: > > > On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote: > >> >> Hello Freeswitch User! >> >> I am using FS since a few weeks. My intent is to have clients who >> uses TLS >> and SRTP for a full encrypted call. >> >> I just managed it, that calls are encrypted with TLS and SRTP. My >> second aim >> ist to redirect this calls for reduce the processing of the server. >> I only >> use the FS for calls between users of this freeswitch. I just tested a >> "redirect" in the dialplan (without TLS and SRTP) but it doesn?t >> work. How i >> have to configure the dialplan for redirect the call to the other >> user. > > You can't. Its not possible because we are a b2bua and you have > already negotiated the keys between the endpoints and FreeSWITCH and > when you redirect the media neither phone can decrypt the packets > correctly. > >> >> I use a SNOM hardphone and a phonerlite softphone. >> >> Thanks for sour help! > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23989162.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From john at feith.com Thu Jun 11 14:17:34 2009 From: john at feith.com (John Wehle) Date: Thu, 11 Jun 2009 17:17:34 -0400 (EDT) Subject: [Freeswitch-users] Caller id when doing transfers Message-ID: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> > It appears from some limited testing that the original caller id is always > shown when the call is transfered. Is there some way to have the person > making the transfer show up as the caller id? To answer my own question it appears that the information is available in the sip_h_Referred-By variable. E.g.: allows the station id making the transfer to be known when a call is transfered to *5. The station id can then be used to park the call in the proper fifo. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From msc at freeswitch.org Thu Jun 11 14:20:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 14:20:13 -0700 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: References: Message-ID: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > > > Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 > sessions remaining like the one below (number and ISP changed) > > > > Anyone have an idea why these 6 sessions remain? I also had 120 calls > that I didn?t get a hang-up for, but that might be me not processing the > events fast enough. > > > Do they show on "show calls"? Or do they show up on "show channels" only? Just curious to see if they were bridged or not. -MC > That said, FS was handling a steady concurrent call level of around 350 > which was awesome !! > > > > Regards > > > > > > UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 > microseconds > > 86913 session(s) since startup > > > > f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 > 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net > ,CS_NEW,,,,,,,,,,,,, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/a27bc751/attachment.html From msc at freeswitch.org Thu Jun 11 14:20:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 14:20:52 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <016601c9eacd$b8ee4cb0$2acae610$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> Message-ID: <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb wrote: > I have a match expression for outbound calls as ?\d{10}?. It?s fine for > unformatted numbers. Not knowing any better, I created another extension to > handle numbers formatted like XXX-XXX-XXXX, which is easier to read and > exists in one hard phone?s phonebook. > > > > It looks like: ?^1?(\d{3})-(\d{3})-(\d{4})$?. But I can see making many > extensions for different formats. > Out of curiosity, what benefit does having all these formats get you? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/1a0f1b8d/attachment.html From msc at freeswitch.org Thu Jun 11 14:21:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 14:21:57 -0700 Subject: [Freeswitch-users] Caller id when doing transfers In-Reply-To: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> References: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> Message-ID: <87f2f3b90906111421u5781c7bbob3704486f745a6cc@mail.gmail.com> On Thu, Jun 11, 2009 at 2:17 PM, John Wehle wrote: > > It appears from some limited testing that the original caller id is > always > > shown when the call is transfered. Is there some way to have the person > > making the transfer show up as the caller id? > > To answer my own question it appears that the information is available > in the sip_h_Referred-By variable. E.g.: > > > > expression="^ > > allows the station id making the transfer to be known when a call is > transfered to *5. The station id can then be used to park the call in > the proper fifo. John, Would you be willing to add this wonderful knowledge to the wiki? :) Let me know if you have any questions about where/how to add it and we'll come up with something. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/302d3747/attachment.html From brian at freeswitch.org Thu Jun 11 14:25:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 16:25:12 -0500 Subject: [Freeswitch-users] re direct calls In-Reply-To: <23989162.post@talk.nabble.com> References: <23982889.post@talk.nabble.com> <23989162.post@talk.nabble.com> Message-ID: <38290DC6-93F2-4104-9A60-951ECE54A68F@freeswitch.org> what? On Jun 11, 2009, at 4:16 PM, NOx-WHV wrote: > Can you just announce b2bua. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/93dc9396/attachment.html From brian at freeswitch.org Thu Jun 11 14:26:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 16:26:29 -0500 Subject: [Freeswitch-users] Caller id when doing transfers In-Reply-To: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> References: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> Message-ID: It does if you do a blind transfer... if you're talking attended transfers thats a whole different ball of wax... /b On Jun 11, 2009, at 4:17 PM, John Wehle wrote: >> It appears from some limited testing that the original caller id is >> always >> shown when the call is transfered. Is there some way to have the >> person >> making the transfer show up as the caller id? > > To answer my own question it appears that the information is available > in the sip_h_Referred-By variable. E.g.: > > > > > > allows the station id making the transfer to be known when a call is > transfered to *5. The station id can then be used to park the call in > the proper fifo. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/aaa89b77/attachment-0001.html From brian at freeswitch.org Thu Jun 11 14:27:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 16:27:58 -0500 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> Message-ID: You could also attach to it with GDB and see if its hanging somewhere else. /b On Jun 11, 2009, at 4:20 PM, Michael Collins wrote: > Do they show on "show calls"? Or do they show up on "show channels" > only? Just curious to see if they were bridged or not. > -MC Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/ed18bc1c/attachment.html From enno.egbert at web.de Thu Jun 11 14:39:40 2009 From: enno.egbert at web.de (NOx-WHV) Date: Thu, 11 Jun 2009 14:39:40 -0700 (PDT) Subject: [Freeswitch-users] re direct calls In-Reply-To: <23989162.post@talk.nabble.com> References: <23982889.post@talk.nabble.com> <23989162.post@talk.nabble.com> Message-ID: <23989451.post@talk.nabble.com> back to back user agent! :-) Thanks! I just ask google! NOx-WHV wrote: > > Thanks for your answer. > > Can you just announce b2bua. > > > > Brian West-3 wrote: >> >> >> On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote: >> >>> >>> Hello Freeswitch User! >>> >>> I am using FS since a few weeks. My intent is to have clients who >>> uses TLS >>> and SRTP for a full encrypted call. >>> >>> I just managed it, that calls are encrypted with TLS and SRTP. My >>> second aim >>> ist to redirect this calls for reduce the processing of the server. >>> I only >>> use the FS for calls between users of this freeswitch. I just tested a >>> "redirect" in the dialplan (without TLS and SRTP) but it doesn?t >>> work. How i >>> have to configure the dialplan for redirect the call to the other >>> user. >> >> You can't. Its not possible because we are a b2bua and you have >> already negotiated the keys between the endpoints and FreeSWITCH and >> when you redirect the media neither phone can decrypt the packets >> correctly. >> >>> >>> I use a SNOM hardphone and a phonerlite softphone. >>> >>> Thanks for sour help! >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23989451.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Jun 11 14:40:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 16:40:17 -0500 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: References: Message-ID: <191c3a030906111440g7c60a82dt1e4eab5ba6e45857@mail.gmail.com> it may be a race in the sql event handler where the delete comes before the insert on a really short call. how many sessions did "status" report were in use? On Thu, Jun 11, 2009 at 4:07 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > > > Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 > sessions remaining like the one below (number and ISP changed) > > > > Anyone have an idea why these 6 sessions remain? I also had 120 calls > that I didn?t get a hang-up for, but that might be me not processing the > events fast enough. > > > > That said, FS was handling a steady concurrent call level of around 350 > which was awesome !! > > > > Regards > > > > > > UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 > microseconds > > 86913 session(s) since startup > > > > f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 > 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net > ,CS_NEW,,,,,,,,,,,,, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/7ba1cfca/attachment.html From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:40:43 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:40:43 +0100 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> Message-ID: It was the output from show channels. I've rebooted the server now, so I can't run show calls. I'll see what happens tomorrow. Certainly running status showed 6 sessions All calls are initiated using and 'Originate' from an inbound socket Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 11 June 2009 22:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton wrote: Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. Do they show on "show calls"? Or do they show up on "show channels" only? Just curious to see if they were bridged or not. -MC That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,, ,,,,, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/d9de3a1c/attachment-0001.html From mike at jerris.com Thu Jun 11 14:50:42 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Jun 2009 17:50:42 -0400 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> Message-ID: <1E7EA5F8-FAFD-40FD-B7CC-22B396DC5B3B@jerris.com> If they were still showing in status, can you use gcore to dump a core next time this happens, leave it running somewhere we can get to it and post a thread apply all bt to Jira. Mike On Jun 11, 2009, at 5:40 PM, "Nik Middleton" wrote: > It was the output from show channels. I?ve rebooted the server now, > so I can?t run show calls. I?ll see what happens tomorrow. Certai > nly running status showed 6 sessions > > > > All calls are initiated using and ?Originate? from an inbound socket > > > > Regards > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: 11 June 2009 22:20 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Orphaned calls > > > > > > On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton > wrote: > > > > Ok, so I did a mere 86,000 calls today, but when it was all over, I > had 6 sessions remaining like the one below (number and ISP changed) > > > > Anyone have an idea why these 6 sessions remain? I also had 120 > calls that I didn?t get a hang-up for, but that might be me not proc > essing the events fast enough. > > > > Do they show on "show calls"? Or do they show up on "show channels" > only? Just curious to see if they were bridged or not. > -MC > > That said, FS was handling a steady concurrent call level of around > 350 which was awesome !! > > > > Regards > > > > > > UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 > milliseconds, 400 microseconds > > 86913 session(s) since startup > > > > f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 > 18:38:23,1244741903,sofia/external/ > 0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,,,,,,, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/3b1b1ea1/attachment.html From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:57:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:57:33 +0100 Subject: [Freeswitch-users] Status Event Message-ID: Not sure where enhancement requests should be posted, but here it is anyway I would dearly love to be able to send a status event that returns an event style output that provides machine readable output rather than the wordy human readable response. (I hate parsing) Is there such an event already? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/30ba894d/attachment.html From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:58:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:58:51 +0100 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: <1E7EA5F8-FAFD-40FD-B7CC-22B396DC5B3B@jerris.com> References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> <1E7EA5F8-FAFD-40FD-B7CC-22B396DC5B3B@jerris.com> Message-ID: Will do Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 11 June 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls If they were still showing in status, can you use gcore to dump a core next time this happens, leave it running somewhere we can get to it and post a thread apply all bt to Jira. Mike On Jun 11, 2009, at 5:40 PM, "Nik Middleton" wrote: It was the output from show channels. I've rebooted the server now, so I can't run show calls. I'll see what happens tomorrow. Certainly running status showed 6 sessions All calls are initiated using and 'Originate' from an inbound socket Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 11 June 2009 22:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. Do they show on "show calls"? Or do they show up on "show channels" only? Just curious to see if they were bridged or not. -MC That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/ 0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,,,,,,, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c51d0134/attachment-0001.html From larclap at yahoo.com Thu Jun 11 15:57:52 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 15:57:52 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> Message-ID: <01fc01c9eae8$0cf317e0$26d947a0$@com> The users entering numbers into their phonebooks are able to recognize the number more easily. I will tell them to forget it and make the phone numbers numeric only. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, June 11, 2009 2:21 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb wrote: I have a match expression for outbound calls as "\d{10}". It's fine for unformatted numbers. Not knowing any better, I created another extension to handle numbers formatted like XXX-XXX-XXXX, which is easier to read and exists in one hard phone's phonebook. It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many extensions for different formats. Out of curiosity, what benefit does having all these formats get you? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/556cec3e/attachment.html From brian at freeswitch.org Thu Jun 11 16:04:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 18:04:57 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <01fc01c9eae8$0cf317e0$26d947a0$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> Message-ID: <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> If the phone sends them with dashes in them the phone IS BROKEN and should be smashed with a hammer. /b On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote: > The users entering numbers into their phonebooks are able to > recognize the number more easily. > > I will tell them to forget it and make the phone numbers numeric only. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/e32cc907/attachment.html From jmesquita at gmail.com Thu Jun 11 16:25:05 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 11 Jun 2009 20:25:05 -0300 Subject: [Freeswitch-users] Status Event In-Reply-To: References: Message-ID: <5a8712120906111625o46598794ica826e07001d4dcc@mail.gmail.com> Nik, I am a noobie and all, but most API responses can come as xml just by adding "as xml" at the end of the call. jmesquita On Thu, Jun 11, 2009 at 6:57 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Not sure where enhancement requests should be posted, but here it is > anyway > > > > > > I would dearly love to be able to send a status event that returns an event > style output that provides machine readable output rather than the wordy > human readable response. (I hate parsing) > > > > Is there such an event already? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/02b5295f/attachment.html From brian at freeswitch.org Thu Jun 11 16:35:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 18:35:00 -0500 Subject: [Freeswitch-users] Status Event In-Reply-To: <5a8712120906111625o46598794ica826e07001d4dcc@mail.gmail.com> References: <5a8712120906111625o46598794ica826e07001d4dcc@mail.gmail.com> Message-ID: <8CE37A09-F0B2-47F5-94AE-CDB74483A1FF@freeswitch.org> Only if they have an as xml modifier /b On Jun 11, 2009, at 6:25 PM, Jo?o Mesquita wrote: > Nik, I am a noobie and all, but most API responses can come as xml > just by adding "as xml" at the end of the call. > > jmesquita Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/4a00a732/attachment.html From larclap at yahoo.com Thu Jun 11 17:30:39 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 17:30:39 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> Message-ID: <021f01c9eaf5$033e2020$09ba6060$@com> It's a SNOM 320. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 4:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match If the phone sends them with dashes in them the phone IS BROKEN and should be smashed with a hammer. /b On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote: The users entering numbers into their phonebooks are able to recognize the number more easily. I will tell them to forget it and make the phone numbers numeric only. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/b1282488/attachment-0001.html From brian at freeswitch.org Thu Jun 11 17:39:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 19:39:43 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <021f01c9eaf5$033e2020$09ba6060$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> Message-ID: <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> What firmware? /b On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote: > It?s a SNOM 320. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c0938532/attachment.html From larclap at yahoo.com Thu Jun 11 18:34:14 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 18:34:14 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> Message-ID: <023301c9eafd$e509f3f0$af1ddbd0$@com> snom320-SIP 6.5.17. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 5:40 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match What firmware? /b On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote: It's a SNOM 320. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/ec2de897/attachment.html From brian at freeswitch.org Thu Jun 11 18:41:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 20:41:12 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <023301c9eafd$e509f3f0$af1ddbd0$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> <023301c9eafd$e509f3f0$af1ddbd0$@com> Message-ID: <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> You should be running 7.1.35 or higher. /b On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote: > snom320-SIP 6.5.17. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/aef7261a/attachment.html From larclap at yahoo.com Thu Jun 11 19:21:46 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 19:21:46 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> <023301c9eafd$e509f3f0$af1ddbd0$@com> <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> Message-ID: <024401c9eb04$89538b00$9bfaa100$@com> snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX-XXXX but delivers XXXXXXXXXX to FS. Thanks Brian From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 6:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match You should be running 7.1.35 or higher. /b On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote: snom320-SIP 6.5.17. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/1a6968d6/attachment-0001.html From brian at freeswitch.org Thu Jun 11 19:27:23 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 21:27:23 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <024401c9eb04$89538b00$9bfaa100$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> <023301c9eafd$e509f3f0$af1ddbd0$@com> <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> <024401c9eb04$89538b00$9bfaa100$@com> Message-ID: See I knew that was a bit of crack :P, Good to hear its working like it SHOULD now! /b On Jun 11, 2009, at 9:21 PM, Lars Zeb wrote: > snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX-XXXX > but delivers XXXXXXXXXX to FS. > > Thanks Brian Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c96ea677/attachment.html From johnd at defyne.org Thu Jun 11 20:35:58 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 13:35:58 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Message-ID: Hi, On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > > Well, if you're running multiple machines, waiting for it to drainstop > isn't that big of a deal unless you're in some sort of hurry, right? > Give it an hour or so to drainstop, then kill 'em. Yes that's exactly what I'm trying to do. The problem is some people will only try one IP address. > Would it not be simpler to try to do something with re-invites or REFER, > assuming your endpoints support it? That was actually plan A. I already added a property in sip_profile called failover_redirect, which specifies another server to try if FS can't allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), by sending back a SIP 302 Moved Temporarily response, instead of 503 Max Calls In Progress. Turns out not all my endpoints support it :( I considered REFER too but there seems to be even less support for that. If I can't get the socket-sharing upgrade working then I will fall back to this - and peers which don't support the 302 response (or more likely, don't authorise it) will just get no service during the upgrade. > -Michael {P^/ > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Dalgliesh > Sent: Thursday, June 11, 2009 12:14 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Live Upgrade Techniques > > > I assume he's talking about hardware failures here :P > > But to answer the question: crashes are easy to deal with. With a crash > you have lost the calls that are in progress anyway; you don't have to > manage a gradual transition. > > Currently, since FS is quite quick to start up, I am just relaunching it > immediately. > > But when I have a second box up and running what I'll do is just add the > IP of the dead machine as another IP of the second box, and then it will > take all the old machine's traffic. That is the plan anyway. I've seen > some commercial boxes that use a similar trick. > ... From brian at freeswitch.org Thu Jun 11 20:57:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 22:57:37 -0500 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Message-ID: <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > > Hi, > > On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >> >> Well, if you're running multiple machines, waiting for it to >> drainstop >> isn't that big of a deal unless you're in some sort of hurry, right? >> Give it an hour or so to drainstop, then kill 'em. > > Yes that's exactly what I'm trying to do. The problem is some people > will > only try one IP address. Clients that don't properly implement SRV/NAPTR and fail over need to be smacked. :) (not customers but software that fails to do that) > >> Would it not be simpler to try to do something with re-invites or >> REFER, >> assuming your endpoints support it? > > That was actually plan A. I already added a property in sip_profile > called > failover_redirect, which specifies another server to try if FS can't > allocate any more sessions (e.g. too busy, paused, shutdown asap, > etc.), > by sending back a SIP 302 Moved Temporarily response, instead of 503 > Max > Calls In Progress. You can't send a 302 to a call thats already established. > > Turns out not all my endpoints support it :( AKA broken endpoints. :) > > I considered REFER too but there seems to be even less support for > that. ACK really? thats sad! > > If I can't get the socket-sharing upgrade working then I will fall > back to > this - and peers which don't support the 302 response (or more likely, > don't authorise it) will just get no service during the upgrade. > >> -Michael > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/04f3ccda/attachment.html From mgg at giagnocavo.net Thu Jun 11 21:16:51 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 12 Jun 2009 00:16:51 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23F60@mse17be1.mse17.exchange.ms> >> Well, if you're running multiple machines, waiting for it to drainstop >> isn't that big of a deal unless you're in some sort of hurry, right? >> Give it an hour or so to drainstop, then kill 'em. > >Yes that's exactly what I'm trying to do. The problem is some people will >only try one IP address. Right, so if you have a proxy in front that is handling this, it should be no problem. From shahzad at vopium.com Thu Jun 11 21:45:57 2009 From: shahzad at vopium.com (Muhammad Shahzad) Date: Fri, 12 Jun 2009 10:45:57 +0600 Subject: [Freeswitch-users] MPL Confusion Message-ID: Hi, I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do understand that me or any one can change provided code according to our customization needs and we are not bound to share our changes as long as we are not distributing it, right? Now, i have been doing R&D on MSN and Yahoo voice chat services, I have now completed by research and now would like to write up FS modules to communicate with these servers. But as you all know both MSN and Yahoo provide SIP based VOIP services, however they are not using standard SIP stack and have their own versions of customized SIP stack. So, in order to write an endpoint for these servers, instead of writing everything from scretch, i can using existing mod_sofia endpoint and customize it to make it compatible with MSN and Yahoo SIP stack. So here are my questions, 1. Is it possible under MPL, that i make a copy of mod_sofia as say mod_msn and develop it to work with MSN, similarly mod_yahoo for Yahoo voice chat service? 2. If yes, how can i mention my role in these modules development, i.e. as developer or as contributor? Also i wish to include my work, once completed, in FreeSWITCH, can you provide me the guidelines and / or eligibility criteria to do so, any link on FS site etc.? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/f9d2d154/attachment-0001.html From johnd at defyne.org Thu Jun 11 23:25:58 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 16:25:58 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> Message-ID: On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >>> >>> Well, if you're running multiple machines, waiting for it to drainstop >>> isn't that big of a deal unless you're in some sort of hurry, right? >>> Give it an hour or so to drainstop, then kill 'em. >> >> Yes that's exactly what I'm trying to do. The problem is some people will >> only try one IP address. > > Clients that don't properly implement SRV/NAPTR and fail over need to be > smacked. :) (not customers but software that fails to do that) Yes I'm sure much of their software can do this but it has been set up for static numeric IPs. And getting the IP changed is a week-long process for some customers! >>> Would it not be simpler to try to do something with re-invites or REFER, >>> assuming your endpoints support it? >> >> That was actually plan A. I already added a property in sip_profile called >> failover_redirect, which specifies another server to try if FS can't >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max >> Calls In Progress. > > You can't send a 302 to a call thats already established. Yes and I don't want to touch established calls - those calls can stay there until they drop. This is sent to new requests when switch_core_session_request fails in mod_sofia. >> Turns out not all my endpoints support it :( > > AKA broken endpoints. :) Some are broken. Some just have this feature disabled. For 'security reasons'. You know the drill. {P^/ John From saeedahmad1981 at gmail.com Fri Jun 12 03:16:27 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 12 Jun 2009 12:16:27 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> Message-ID: I've experience with a commercial SBC, these are two machines running in cluster mode. In that case if one SBC is going down then other will take all new calls including the call which were active on broken SBC (SIP only). Thats quite ideal for wholesale traffic where the SBC will never be idle. On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh wrote: > On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > >>> > >>> Well, if you're running multiple machines, waiting for it to drainstop > >>> isn't that big of a deal unless you're in some sort of hurry, right? > >>> Give it an hour or so to drainstop, then kill 'em. > >> > >> Yes that's exactly what I'm trying to do. The problem is some people > will > >> only try one IP address. > > > > Clients that don't properly implement SRV/NAPTR and fail over need to be > > smacked. :) (not customers but software that fails to do that) > > Yes I'm sure much of their software can do this but it has been set up for > static numeric IPs. And getting the IP changed is a week-long process for > some customers! > > >>> Would it not be simpler to try to do something with re-invites or > REFER, > >>> assuming your endpoints support it? > >> > >> That was actually plan A. I already added a property in sip_profile > called > >> failover_redirect, which specifies another server to try if FS can't > >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), > >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max > >> Calls In Progress. > > > > You can't send a 302 to a call thats already established. > > Yes and I don't want to touch established calls - those calls can stay > there until they drop. This is sent to new requests when > switch_core_session_request fails in mod_sofia. > > >> Turns out not all my endpoints support it :( > > > > AKA broken endpoints. :) > > Some are broken. Some just have this feature disabled. For 'security > reasons'. You know the drill. > > > {P^/ > John > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/b7f3161b/attachment.html From helmut.kuper at ewetel.de Fri Jun 12 05:43:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 12 Jun 2009 14:43:02 +0200 Subject: [Freeswitch-users] Database and "Too many open files" Problem In-Reply-To: <191c3a030906101414j1107e49bk3aa8fb778454451c@mail.gmail.com> References: <4A2FEE39.1030709@ewetel.de> <191c3a030906101414j1107e49bk3aa8fb778454451c@mail.gmail.com> Message-ID: <4A324D56.6080304@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Anthony, yes, that's true. Works well now. Thanks a lot! regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKMk1W4tZeNddg3dwRAhzrAJ45OpGkPkpLEPRw17HUpR3CTaxVVwCcD4/0 TJpI0jZez6uOdETu3OtDbc8= =yWhz -----END PGP SIGNATURE----- From grevenx at me.com Fri Jun 12 06:03:33 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Fri, 12 Jun 2009 15:03:33 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> Message-ID: <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Can you comment some more on how this is configured? Would it be something that could be added to the wiki in the SBC setup page? Best regards, Even Andr? Fiskvik On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: > I've experience with a commercial SBC, these are two machines > running in cluster mode. In that case if one SBC is going down then > other will take all new calls including the call which were active > on broken SBC (SIP only). > > Thats quite ideal for wholesale traffic where the SBC will never be > idle. > > On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh > wrote: > On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > >>> > >>> Well, if you're running multiple machines, waiting for it to > drainstop > >>> isn't that big of a deal unless you're in some sort of hurry, > right? > >>> Give it an hour or so to drainstop, then kill 'em. > >> > >> Yes that's exactly what I'm trying to do. The problem is some > people will > >> only try one IP address. > > > > Clients that don't properly implement SRV/NAPTR and fail over need > to be > > smacked. :) (not customers but software that fails to do that) > > Yes I'm sure much of their software can do this but it has been set > up for > static numeric IPs. And getting the IP changed is a week-long > process for > some customers! > > >>> Would it not be simpler to try to do something with re-invites > or REFER, > >>> assuming your endpoints support it? > >> > >> That was actually plan A. I already added a property in > sip_profile called > >> failover_redirect, which specifies another server to try if FS > can't > >> allocate any more sessions (e.g. too busy, paused, shutdown asap, > etc.), > >> by sending back a SIP 302 Moved Temporarily response, instead of > 503 Max > >> Calls In Progress. > > > > You can't send a 302 to a call thats already established. > > Yes and I don't want to touch established calls - those calls can stay > there until they drop. This is sent to new requests when > switch_core_session_request fails in mod_sofia. > > >> Turns out not all my endpoints support it :( > > > > AKA broken endpoints. :) > > Some are broken. Some just have this feature disabled. For 'security > reasons'. You know the drill. > > > {P^/ > John > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/6fec5bb1/attachment.html From larclap at yahoo.com Fri Jun 12 06:33:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 12 Jun 2009 06:33:12 -0700 Subject: [Freeswitch-users] Unregister extension? Message-ID: <003901c9eb62$551d0f60$ff572e20$@com> How can I unregister a softphone's registration? I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I changed the second one to 1000. Now when I do 'sofia status profile internal' all three show up. How do I get rid of the 1001 extension? I shutdown and restarted FS but that didn't do it. I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is blocking the Polycom at that same extension and that is the reason the Polycom is not showing. Thanks, Lars Registrations: ============================================================================ ===================== Call-ID: 3c267015ab6b-bd6gioq5ytor User: 1010 at 192.168.10.29 Contact: "1010" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1010 Auth-Realm: 192.168.10.29 Call-ID: 3c267015afa6-6v0sw4o3qei3 User: 1001 at 192.168.10.29 Contact: "1001" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1001 Auth-Realm: 192.168.10.29 Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. User: 1019 at 192.168.10.29 Contact: "1019" Agent: Bria Professional release 2.4.3 stamp 50906 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) Host: fs IP: 192.168.10.11 Port: 19040 Auth-User: 1019 Auth-Realm: 192.168.10.29 Call-ID: 3c270d667ff5-47fq2p6n1ou1 User: 1000 at 192.168.10.29 Contact: "1000" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1000 Auth-Realm: 192.168.10.29 ============================================================================ ===================== -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/fea90e20/attachment-0001.html From saeedahmad1981 at gmail.com Fri Jun 12 06:39:27 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 12 Jun 2009 15:39:27 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms><15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca><6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Message-ID: No idea at all, It?s a commercial SBC. I wish if we can have same functionality in FS. - Saeed _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Even Andr? Fiskvik Sent: Friday, June 12, 2009 3:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques Can you comment some more on how this is configured? Would it be something that could be added to the wiki in the SBC setup page? Best regards, Even Andr? Fiskvik On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: I've experience with a commercial SBC, these are two machines running in cluster mode. In that case if one SBC is going down then other will take all new calls including the call which were active on broken SBC (SIP only). Thats quite ideal for wholesale traffic where the SBC will never be idle. On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh wrote: On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >>> >>> Well, if you're running multiple machines, waiting for it to drainstop >>> isn't that big of a deal unless you're in some sort of hurry, right? >>> Give it an hour or so to drainstop, then kill 'em. >> >> Yes that's exactly what I'm trying to do. The problem is some people will >> only try one IP address. > > Clients that don't properly implement SRV/NAPTR and fail over need to be > smacked. :) (not customers but software that fails to do that) Yes I'm sure much of their software can do this but it has been set up for static numeric IPs. And getting the IP changed is a week-long process for some customers! >>> Would it not be simpler to try to do something with re-invites or REFER, >>> assuming your endpoints support it? >> >> That was actually plan A. I already added a property in sip_profile called >> failover_redirect, which specifies another server to try if FS can't >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max >> Calls In Progress. > > You can't send a 302 to a call thats already established. Yes and I don't want to touch established calls - those calls can stay there until they drop. This is sent to new requests when switch_core_session_request fails in mod_sofia. >> Turns out not all my endpoints support it :( > > AKA broken endpoints. :) Some are broken. Some just have this feature disabled. For 'security reasons'. You know the drill. {P^/ John _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/ffb2220c/attachment.html From jmesquita at gmail.com Fri Jun 12 06:47:12 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 12 Jun 2009 10:47:12 -0300 Subject: [Freeswitch-users] Unregister extension? In-Reply-To: <003901c9eb62$551d0f60$ff572e20$@com> References: <003901c9eb62$551d0f60$ff572e20$@com> Message-ID: <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> Lars, don't get me wrong but you have been asking questions that are all answered on the wiki: http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoints Might be a good idea to value the work of lots of ppl who have been documenting by actually using the documentation, no? Sorry if that sounds a bit harsh. jmesquita On Fri, Jun 12, 2009 at 10:33 AM, Lars Zeb wrote: > How can I unregister a softphone?s registration? > > > > I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I > changed the second one to 1000. Now when I do ?sofia status profile > internal? all three show up. How do I get rid of the 1001 extension? I > shutdown and restarted FS but that didn?t do it. > > > > I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is > blocking the Polycom at that same extension and that is the reason the > Polycom is not showing. > > > > Thanks, Lars > > > > > > Registrations: > > > ================================================================================================= > > Call-ID: 3c267015ab6b-bd6gioq5ytor > > User: 1010 at 192.168.10.29 > > Contact: "1010" > > Agent: snom320/7.3.14 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) > > Host: fs > > IP: 192.168.10.104 > > Port: 2048 > > Auth-User: 1010 > > Auth-Realm: 192.168.10.29 > > > > Call-ID: 3c267015afa6-6v0sw4o3qei3 > > User: 1001 at 192.168.10.29 > > Contact: "1001" > > Agent: snom320/7.3.14 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) > > Host: fs > > IP: 192.168.10.104 > > Port: 2048 > > Auth-User: 1001 > > Auth-Realm: 192.168.10.29 > > > > Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. > > User: 1019 at 192.168.10.29 > > Contact: "1019" ;rinstance=5394acb4dfa00c0a> > > Agent: Bria Professional release 2.4.3 stamp 50906 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) > > Host: fs > > IP: 192.168.10.11 > > Port: 19040 > > Auth-User: 1019 > > Auth-Realm: 192.168.10.29 > > > > Call-ID: 3c270d667ff5-47fq2p6n1ou1 > > User: 1000 at 192.168.10.29 > > Contact: "1000" > > Agent: snom320/7.3.14 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) > > Host: fs > > IP: 192.168.10.104 > > Port: 2048 > > Auth-User: 1000 > > Auth-Realm: 192.168.10.29 > > > > > ================================================================================================= > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/58ea5b09/attachment-0001.html From larclap at yahoo.com Fri Jun 12 07:07:52 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 12 Jun 2009 07:07:52 -0700 Subject: [Freeswitch-users] Unregister extension? In-Reply-To: <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> References: <003901c9eb62$551d0f60$ff572e20$@com> <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> Message-ID: <005801c9eb67$2d3c1180$87b43480$@com> No, it?s not too harsh, Jo?o, but I hope not all of my questions were answered on the wiki. I do try to go to the wiki first. I think that my total ignorance of the environment makes it difficult for me to do a search on the wiki or Google. I did try before asking this list. My query to Google was ?Freeswitch unregister?. That was the best I could do given my limited knowledge. Thank you for the help. I?ll learn eventually. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Friday, June 12, 2009 6:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Unregister extension? Lars, don't get me wrong but you have been asking questions that are all answered on the wiki: http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoi nts Might be a good idea to value the work of lots of ppl who have been documenting by actually using the documentation, no? Sorry if that sounds a bit harsh. jmesquita On Fri, Jun 12, 2009 at 10:33 AM, Lars Zeb wrote: How can I unregister a softphone?s registration? I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I changed the second one to 1000. Now when I do ?sofia status profile internal? all three show up. How do I get rid of the 1001 extension? I shutdown and restarted FS but that didn?t do it. I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is blocking the Polycom at that same extension and that is the reason the Polycom is not showing. Thanks, Lars Registrations: ============================================================================ ===================== Call-ID: 3c267015ab6b-bd6gioq5ytor User: 1010 at 192.168.10.29 Contact: "1010" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1010 Auth-Realm: 192.168.10.29 Call-ID: 3c267015afa6-6v0sw4o3qei3 User: 1001 at 192.168.10.29 Contact: "1001" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1001 Auth-Realm: 192.168.10.29 Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. User: 1019 at 192.168.10.29 Contact: "1019" Agent: Bria Professional release 2.4.3 stamp 50906 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) Host: fs IP: 192.168.10.11 Port: 19040 Auth-User: 1019 Auth-Realm: 192.168.10.29 Call-ID: 3c270d667ff5-47fq2p6n1ou1 User: 1000 at 192.168.10.29 Contact: "1000" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1000 Auth-Realm: 192.168.10.29 ============================================================================ ===================== _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/a67f8ac4/attachment.html From helmut.kuper at ewetel.de Fri Jun 12 07:32:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 12 Jun 2009 16:32:05 +0200 Subject: [Freeswitch-users] Q931 TE State Timer Message-ID: <4A3266E5.2000702@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I just want to let you know, that I started some work on Q.931 TE state timers in current openzap ozmod_isdn stack. The current stack has some problems with freeing ressources when far end doesn't follow Q931 state machine cleanly. On my side I mainly miss a RELEASE after sending DISCONNECT and so m more and more channels are wasted over time. Openzap Q931 stack has some preworked timer and state handling (in form of Q931_UX states). Unfortunately it is disabled in source. I enabled it, added some Q931_UX states to Q931StateTE.c, added a timeout handler to ozmod_isdn.c and tested it with my special problem. In lab missed RELEASEs are now detected and the corresponding channel is freed cleanly - - hurray :) I would like to put my work in openzap trunk as soon as it works stable in production and FS board allows me to do so. I can't promise that I will implement all timers for Q931 TE nor I plan to work on Q931 NT timers. As I said: Just for your information. Have a nice weekend! Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKMmbl4tZeNddg3dwRAvWbAJ43Vex8J3LiVMu9mvs4US23eif19gCeO37l 4CGbk8CA+n+/dMRo5y5A+i0= =ZtvM -----END PGP SIGNATURE----- From andy at fabulous4.co.uk Fri Jun 12 07:37:39 2009 From: andy at fabulous4.co.uk (Andy) Date: Fri, 12 Jun 2009 15:37:39 +0100 Subject: [Freeswitch-users] Sample rate and recordFile Message-ID: <137853E7923C47B7890E39796657719E@D810> Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/a3f1625c/attachment.html From brian at freeswitch.org Fri Jun 12 07:44:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 09:44:54 -0500 Subject: [Freeswitch-users] MPL Confusion In-Reply-To: References: Message-ID: <8DBC35F3-0ADE-4391-BEEB-99609F02A739@freeswitch.org> On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: > Hi, > > I have some confusion about FreeSWITCH's Mozilla Public License 1.1. > I do understand that me or any one can change provided code > according to our customization needs and we are not bound to share > our changes as long as we are not distributing it, right? Not 100% true. > > Now, i have been doing R&D on MSN and Yahoo voice chat services, I > have now completed by research and now would like to write up FS > modules to communicate with these servers. But as you all know both > MSN and Yahoo provide SIP based VOIP services, however they are not > using standard SIP stack and have their own versions of customized > SIP stack. So, in order to write an endpoint for these servers, > instead of writing everything from scretch, i can using existing > mod_sofia endpoint and customize it to make it compatible with MSN > and Yahoo SIP stack. So here are my questions, > > 1. Is it possible under MPL, that i make a copy of mod_sofia as say > mod_msn and develop it to work with MSN, similarly mod_yahoo for > Yahoo voice chat service? Chances are they can be integrated as optional behaviors into mod_sofia. Its best to join #freeswitch and talk to us to see if maybe we can provide you guidance on this process. > 2. If yes, how can i mention my role in these modules development, > i.e. as developer or as contributor? Adding your name to the top of the files is usually the best way. > > Also i wish to include my work, once completed, in FreeSWITCH, can > you provide me the guidelines and / or eligibility criteria to do > so, any link on FS site etc.? > You post your work to our issue tracker http://jira.freeswitch.org > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/bd71a6df/attachment-0001.html From brian at freeswitch.org Fri Jun 12 07:49:12 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 09:49:12 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <137853E7923C47B7890E39796657719E@D810> References: <137853E7923C47B7890E39796657719E@D810> Message-ID: <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> No its not possible yet. Voicemail has an option to record in 11025 but not arbitrary rates for recording otherwise. /b On Jun 12, 2009, at 9:37 AM, Andy wrote: > Hi, > > Sorry but I just can't find this in the documentation. I'm using > recordFile to record incoming messages. I'd like the audio files > produced to be 11025Hz rather than 8kHz is this possible? What > setting do I need to change? > > Many thanks > Andy > ________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/de8767b8/attachment.html From brian at freeswitch.org Fri Jun 12 07:51:02 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 09:51:02 -0500 Subject: [Freeswitch-users] Unregister extension? In-Reply-To: <005801c9eb67$2d3c1180$87b43480$@com> References: <003901c9eb62$551d0f60$ff572e20$@com> <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> <005801c9eb67$2d3c1180$87b43480$@com> Message-ID: Its one of the 3000 settings you can change on the identity ... but in your case I am not sure it would have removed the old registration before the new one was registered... but check on the preferences or the identity there is a setting to unregister on reboot. /b PS: sofia profile xxx flush_inbound_reg [call-id] On Jun 12, 2009, at 9:07 AM, Lars Zeb wrote: > No, it?s not too harsh, Jo?o, but I hope not all of my questions > were answered on the wiki. > > I do try to go to the wiki first. I think that my total ignorance of > the environment makes it difficult for me to do a search on the wiki > or Google. I did try before asking this list. My query to Google was > ?Freeswitch unregister?. That was the best I could do given my > limited knowledge. > > Thank you for the help. I?ll learn eventually. > > Lars > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/2262e0e3/attachment.html From santhosh.suthakar at gmail.com Fri Jun 12 00:35:07 2009 From: santhosh.suthakar at gmail.com (Santhosh) Date: Fri, 12 Jun 2009 00:35:07 -0700 Subject: [Freeswitch-users] XML-RPC In-Reply-To: <4a31fa03.14b48c0a.4983.fffff107@mx.google.com> References: <4a31fa03.14b48c0a.4983.fffff107@mx.google.com> Message-ID: <4a320541.25578c0a.1cf4.0f2d@mx.google.com> Hi, Is there any where I can find more documentation on the XML-RPC interface of freeswitch. I am trying to initiate a conference and add in users to it from flex. I would appreciate any insights. Thanks San -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/6aace86e/attachment.html From mike at jerris.com Fri Jun 12 08:34:26 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 12 Jun 2009 11:34:26 -0400 Subject: [Freeswitch-users] XML-RPC In-Reply-To: <4a320541.25578c0a.1cf4.0f2d@mx.google.com> References: <4a31fa03.14b48c0a.4983.fffff107@mx.google.com> <4a320541.25578c0a.1cf4.0f2d@mx.google.com> Message-ID: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC http://wiki.freeswitch.org/wiki/Mod_commands http://wiki.freeswitch.org/wiki/Mod_conference Mike On Jun 12, 2009, at 3:35 AM, Santhosh wrote: > Hi, > > Is there any where I can find more documentation on the XML-RPC > interface of freeswitch. I am trying to initiate a conference and > add in users to it from flex. I would appreciate any insights. > > Thanks > San > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/4551e9cd/attachment-0001.html From anthony.minessale at gmail.com Fri Jun 12 08:50:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 12 Jun 2009 10:50:43 -0500 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A3266E5.2000702@ewetel.de> References: <4A3266E5.2000702@ewetel.de> Message-ID: <191c3a030906120850q23b816e0qe8be6f7ae9ca8cc5@mail.gmail.com> please make sure you stay tuned into #openzap and coordinate with stkn and the other guys doing work on the stack. That way we can make sure we get the best out of the code. On Fri, Jun 12, 2009 at 9:32 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I just want to let you know, that I started some work on Q.931 TE state > timers in current openzap ozmod_isdn stack. The current stack has some > problems with freeing ressources when far end doesn't follow Q931 state > machine cleanly. On my side I mainly miss a RELEASE after sending > DISCONNECT and so m more and more channels are wasted over time. > > Openzap Q931 stack has some preworked timer and state handling (in form > of Q931_UX states). Unfortunately it is disabled in source. I enabled > it, added some Q931_UX states to Q931StateTE.c, added a timeout handler > to ozmod_isdn.c and tested it with my special problem. In lab missed > RELEASEs are now detected and the corresponding channel is freed cleanly > - - hurray :) > > I would like to put my work in openzap trunk as soon as it works stable > in production and FS board allows me to do so. > > I can't promise that I will implement all timers for Q931 TE nor I plan > to work on Q931 NT timers. > > As I said: Just for your information. > > > Have a nice weekend! > > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKMmbl4tZeNddg3dwRAvWbAJ43Vex8J3LiVMu9mvs4US23eif19gCeO37l > 4CGbk8CA+n+/dMRo5y5A+i0= > =ZtvM > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/c86dca0a/attachment.html From anthony.minessale at gmail.com Fri Jun 12 09:02:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 12 Jun 2009 11:02:57 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> Message-ID: <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> Looked easy enough so i added record_sample_rate variable that should influence it for you if you set it in advance. On Fri, Jun 12, 2009 at 9:49 AM, Brian West wrote: > No its not possible yet. Voicemail has an option to record in 11025 but > not arbitrary rates for recording otherwise. > /b > > On Jun 12, 2009, at 9:37 AM, Andy wrote: > > Hi, > > Sorry but I just can't find this in the documentation. I'm using recordFile > to record incoming messages. I'd like the audio files produced to be 11025Hz > rather than 8kHz is this possible? What setting do I need to change? > > Many thanks > Andy > ________ > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/503b3f7e/attachment.html From andy at fabulous4.co.uk Fri Jun 12 09:26:08 2009 From: andy at fabulous4.co.uk (Andy) Date: Fri, 12 Jun 2009 17:26:08 +0100 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> Message-ID: Excellent, thanks Anthony, I'll give it a go. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 12 June 2009 17:03 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile Looked easy enough so i added record_sample_rate variable that should influence it for you if you set it in advance. On Fri, Jun 12, 2009 at 9:49 AM, Brian West wrote: No its not possible yet. Voicemail has an option to record in 11025 but not arbitrary rates for recording otherwise. /b On Jun 12, 2009, at 9:37 AM, Andy wrote: Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy ________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/882e9c45/attachment.html From mgg at giagnocavo.net Fri Jun 12 09:54:04 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 12 Jun 2009 12:54:04 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms><15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca><6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> Well, Nextone for instance has a database the keeps most of the state of calls, and it's replicated between the two nodes. (I seem to recall the database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still get truncated when there's any kind of switchover. But Nextone is a closed system with limited services. As MikeJ mentioned, it was discussed for FS, but it's a LOT of work to get that state synchronized. And, every custom app/module would have to register and support recreating their state. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Saeed Ahmed Sent: Friday, June 12, 2009 7:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques No idea at all, It's a commercial SBC. I wish if we can have same functionality in FS. - Saeed ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Even Andr? Fiskvik Sent: Friday, June 12, 2009 3:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques Can you comment some more on how this is configured? Would it be something that could be added to the wiki in the SBC setup page? Best regards, Even Andr? Fiskvik On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: I've experience with a commercial SBC, these are two machines running in cluster mode. In that case if one SBC is going down then other will take all new calls including the call which were active on broken SBC (SIP only). Thats quite ideal for wholesale traffic where the SBC will never be idle. On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh > wrote: On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >>> >>> Well, if you're running multiple machines, waiting for it to drainstop >>> isn't that big of a deal unless you're in some sort of hurry, right? >>> Give it an hour or so to drainstop, then kill 'em. >> >> Yes that's exactly what I'm trying to do. The problem is some people will >> only try one IP address. > > Clients that don't properly implement SRV/NAPTR and fail over need to be > smacked. :) (not customers but software that fails to do that) Yes I'm sure much of their software can do this but it has been set up for static numeric IPs. And getting the IP changed is a week-long process for some customers! >>> Would it not be simpler to try to do something with re-invites or REFER, >>> assuming your endpoints support it? >> >> That was actually plan A. I already added a property in sip_profile called >> failover_redirect, which specifies another server to try if FS can't >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max >> Calls In Progress. > > You can't send a 302 to a call thats already established. Yes and I don't want to touch established calls - those calls can stay there until they drop. This is sent to new requests when switch_core_session_request fails in mod_sofia. >> Turns out not all my endpoints support it :( > > AKA broken endpoints. :) Some are broken. Some just have this feature disabled. For 'security reasons'. You know the drill. {P^/ John _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/56e045da/attachment-0001.html From stkn at freeswitch.org Fri Jun 12 10:21:18 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 12 Jun 2009 19:21:18 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A3266E5.2000702@ewetel.de> References: <4A3266E5.2000702@ewetel.de> Message-ID: <4A328E8E.6030607@freeswitch.org> Helmut Kuper wrote: > Hello, > > I just want to let you know, that I started some work on Q.931 TE state > timers in current openzap ozmod_isdn stack. The current stack has some > problems with freeing ressources when far end doesn't follow Q931 state > machine cleanly. On my side I mainly miss a RELEASE after sending > DISCONNECT and so m more and more channels are wasted over time. > > Openzap Q931 stack has some preworked timer and state handling (in form > of Q931_UX states). Unfortunately it is disabled in source. I enabled > it, added some Q931_UX states to Q931StateTE.c, added a timeout handler > to ozmod_isdn.c and tested it with my special problem. In lab missed > RELEASEs are now detected and the corresponding channel is freed cleanly > - hurray :) > > I would like to put my work in openzap trunk as soon as it works stable > in production and FS board allows me to do so. > > I can't promise that I will implement all timers for Q931 TE nor I plan > to work on Q931 NT timers. > > As I said: Just for your information. > > > Have a nice weekend! > > Helmut Umm, you've been doing duplicate work then. The version of ozmod_isdn i have been working on is completely stateful and has a couple of timers already implemented. And i remember giving you the location of the git repository on IRC, earlier this year. (But never got any feedback) stkn _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Fri Jun 12 10:52:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 10:52:09 -0700 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> Message-ID: <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> On Fri, Jun 12, 2009 at 9:26 AM, Andy wrote: > Excellent, thanks Anthony, I'll give it a go. > > Andy, can you report back on your success with this variable? Also, we would appreciate it if you could add an entry to the wiki on the channel_variables page. Let me know if you have any questions and I'll be glad to help. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/9e0a8ff3/attachment.html From intralanman at freeswitch.org Fri Jun 12 11:10:33 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 12 Jun 2009 14:10:33 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms><15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca><6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Message-ID: <4A329A19.4010306@freeswitch.org> Saeed Ahmed wrote: > > No idea at all, > > It's a commercial SBC. > > I wish if we can have same functionality in FS. > You could accomplish parts of this with hearbeat and ldirectord.... the in-session calls aren't going to go anywhere, but if the server crashes, the second one can take over the ip of the first easily enough. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/2f9530a2/attachment.html From msc at freeswitch.org Fri Jun 12 11:22:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 11:22:03 -0700 Subject: [Freeswitch-users] INFO: Important story you need to Digg and read right away Message-ID: <87f2f3b90906121122v6599ffbdxa061aab899ff15fe@mail.gmail.com> Gang, There have been crazy rumors flying around. Tony sets the record straight. Please go here now and digg this story: http://digg.com/software/Anthony_Minessale_of_FreeSWITCH_Discusses_Barracuda_Rumors Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/9b8b8742/attachment.html From benj at teliax.com Fri Jun 12 11:59:14 2009 From: benj at teliax.com (Ben Jones) Date: Fri, 12 Jun 2009 12:59:14 -0600 Subject: [Freeswitch-users] RFC2833 double-digits Message-ID: <4A32A582.6040700@teliax.com> Hi all, We're running into a problem with rfc2833. Here's the situation: A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. The FS console shows it receives the digits once. We send the call to a tier-1 carrier The digits are played read back, but often the first several digits are repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when the user entered '1234.' When I was testing this recently, the call was sent to a Cisco gateway, which the documentation* states has problems with rfc2833, just like Sonus gateways, although that should be fixed past revision 10744. When using inband, everything works great. We are using start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers is inband. My initial thought was that the tier-1 was receiving rfc2833 and inband, but it's not consistent. I'm sure more information will be requested of me to help troubleshoot, so please let me know. Any advice would be very much appreciated. *http://wiki.freeswitch.org/wiki/RTP_Issues -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From brian at freeswitch.org Fri Jun 12 12:08:21 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 14:08:21 -0500 Subject: [Freeswitch-users] RFC2833 double-digits In-Reply-To: <4A32A582.6040700@teliax.com> References: <4A32A582.6040700@teliax.com> Message-ID: <6B15EE47-6F85-4DC1-A3FB-928508D4E11C@freeswitch.org> What device are you using? RTP traces, debug logs something to see what might be taking place.?!?! /b On Jun 12, 2009, at 1:59 PM, Ben Jones wrote: > Hi all, > > We're running into a problem with rfc2833. Here's the situation: > > A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. > The FS console shows it receives the digits once. > We send the call to a tier-1 carrier > The digits are played read back, but often the first several digits > are > repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when > the user entered '1234.' > > When I was testing this recently, the call was sent to a Cisco > gateway, > which the documentation* states has problems with rfc2833, just like > Sonus gateways, although that should be fixed past revision 10744. > > When using inband, everything works great. We are using > start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers > is inband. My initial thought was that the tier-1 was receiving > rfc2833 > and inband, but it's not consistent. > > I'm sure more information will be requested of me to help > troubleshoot, > so please let me know. Any advice would be very much appreciated. > > *http://wiki.freeswitch.org/wiki/RTP_Issues Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/89a8231e/attachment.html From benj at teliax.com Fri Jun 12 12:37:16 2009 From: benj at teliax.com (Ben Jones) Date: Fri, 12 Jun 2009 13:37:16 -0600 Subject: [Freeswitch-users] RFC2833 double-digits In-Reply-To: <6B15EE47-6F85-4DC1-A3FB-928508D4E11C@freeswitch.org> References: <4A32A582.6040700@teliax.com> <6B15EE47-6F85-4DC1-A3FB-928508D4E11C@freeswitch.org> Message-ID: <4A32AE6C.2030007@teliax.com> Testing was done with SJphone for Mac, dtmfmode rfc2833 pt 101. Hopefully this debug log can help: http://pastebin.freeswitch.org/9374 If I need to add, change, whatever, let me know. Thanks for the help. -benj Brian West wrote: > What device are you using? RTP traces, debug logs something to see what > might be taking place.?!?! > > /b > > On Jun 12, 2009, at 1:59 PM, Ben Jones wrote: > >> Hi all, >> >> We're running into a problem with rfc2833. Here's the situation: >> >> A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. >> The FS console shows it receives the digits once. >> We send the call to a tier-1 carrier >> The digits are played read back, but often the first several digits are >> repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when >> the user entered '1234.' >> >> When I was testing this recently, the call was sent to a Cisco gateway, >> which the documentation* states has problems with rfc2833, just like >> Sonus gateways, although that should be fixed past revision 10744. >> >> When using inband, everything works great. We are using >> start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers >> is inband. My initial thought was that the tier-1 was receiving rfc2833 >> and inband, but it's not consistent. >> >> I'm sure more information will be requested of me to help troubleshoot, >> so please let me know. Any advice would be very much appreciated. >> >> *http://wiki.freeswitch.org/wiki/RTP_Issues > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From herb at powercom.com Fri Jun 12 15:09:10 2009 From: herb at powercom.com (Herb Levitin) Date: Fri, 12 Jun 2009 15:09:10 -0700 Subject: [Freeswitch-users] Need to hire an experienced FreeSwitch Developer Message-ID: <026701c9ebaa$6a8d90c0$3fa8b240$@com> I need to hire an experienced FreeSwitch developer to build a small chat bridge that supports VoIP and 1-4 PRI's of TDM. Please contact me at herb at powercom.com or (805)845-8906. From apt.get at gmail.com Fri Jun 12 15:14:42 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 12 Jun 2009 16:14:42 -0600 Subject: [Freeswitch-users] high latency Message-ID: Greetings, I'm new to freeswitch, just playing with it in a home/small office at present. Overall I've been really impressed with it. One issue we've noticed is a pronounced latency on some or perhaps all calls over roughly 500 seconds in duration. The latency is approximately 3 seconds or more round trip. I tried setting jitterbuffer_msec=20 but it has not prevented the problem. The call quality is otherwise good with no noticeable choppiness or jitter. Other local network traffic appears to be irrelevant, as in the latency seems to occur even when the LAN is idle or experiencing only sporadic bursts of activity, like typical browsing. I read the FAQ and searched jitter, lag and latency in the wiki and list archive but didn't come up with anything. Is this a known issue? Could it be that the problem is specific to my platform? I'm running the freeswitch package on pfsense (FreeBSD-based) and nobody in the pfsense forums seems to know the cause or solution, although others have reported the same issue. Much thanks for your input, db From msc at freeswitch.org Fri Jun 12 15:33:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 15:33:54 -0700 Subject: [Freeswitch-users] high latency In-Reply-To: References: Message-ID: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> Can you describe your setup? Need to know what kind of OS and hardware is running FS as well as what kind of phones. Any NAT involved? -MC On Fri, Jun 12, 2009 at 3:14 PM, David Burgess wrote: > Greetings, > > I'm new to freeswitch, just playing with it in a home/small office at > present. Overall I've been really impressed with it. > > One issue we've noticed is a pronounced latency on some or perhaps all > calls over roughly 500 seconds in duration. The latency is > approximately 3 seconds or more round trip. I tried setting > jitterbuffer_msec=20 but it has not prevented the problem. The call > quality is otherwise good with no noticeable choppiness or jitter. > Other local network traffic appears to be irrelevant, as in the > latency seems to occur even when the LAN is idle or experiencing only > sporadic bursts of activity, like typical browsing. > > I read the FAQ and searched jitter, lag and latency in the wiki and > list archive but didn't come up with anything. Is this a known issue? > Could it be that the problem is specific to my platform? I'm running > the freeswitch package on pfsense (FreeBSD-based) and nobody in the > pfsense forums seems to know the cause or solution, although others > have reported the same issue. > > Much thanks for your input, > > db > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/65872f1b/attachment.html From apt.get at gmail.com Fri Jun 12 15:53:57 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 12 Jun 2009 16:53:57 -0600 Subject: [Freeswitch-users] high latency In-Reply-To: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> Message-ID: On Fri, Jun 12, 2009 at 4:33 PM, Michael Collins wrote: > Can you describe your setup? Need to know what kind of OS and hardware is > running FS as well as what kind of phones. Any NAT involved? > -MC FS is running inside pfsense, which is a freeBSD-based firewall (pfsense.org). Hardware is a lightly loaded Soekris net5501 (500 MHz Geode x86, 512 MB RAM)(www.soekris.com). FS external and internal profiles are both listening on WAN with a public IP address. My extensions are 2 lines on a Linksys PAP2T residing on the LAN, registered to the internal profile on the WAN interface. All calls are made via a sip trunk. PAP2T normally reports a decode latency in the neighborhood of 30 ms and jitter of 5 ms. Ping time to my provider's rtp servers is ~42 ms. Every call starts out with imperceptible latency, but at some point the caller notices a long delay, as I said, usually around 3 seconds, and usually only on calls lasting 500 seconds or more. Anything else I can provide? db From msc at freeswitch.org Fri Jun 12 16:28:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 16:28:15 -0700 Subject: [Freeswitch-users] high latency In-Reply-To: References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> Message-ID: <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> On Fri, Jun 12, 2009 at 3:53 PM, David Burgess wrote: > On Fri, Jun 12, 2009 at 4:33 PM, Michael Collins > wrote: > > Can you describe your setup? Need to know what kind of OS and hardware is > > running FS as well as what kind of phones. Any NAT involved? > > -MC > > FS is running inside pfsense, which is a freeBSD-based firewall > (pfsense.org). Hardware is a lightly loaded Soekris net5501 (500 MHz > Geode x86, 512 MB RAM)(www.soekris.com). > > FS external and internal profiles are both listening on WAN with a > public IP address. My extensions are 2 lines on a Linksys PAP2T > residing on the LAN, registered to the internal profile on the WAN > interface. All calls are made via a sip trunk. > > PAP2T normally reports a decode latency in the neighborhood of 30 ms > and jitter of 5 ms. Ping time to my provider's rtp servers is ~42 ms. > > Every call starts out with imperceptible latency, but at some point > the caller notices a long delay, as I said, usually around 3 seconds, > and usually only on calls lasting 500 seconds or more. > > Anything else I can provide? Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen anything like this with FS+pfSense. The FS devs aren't exactly keen on FS + FBSD because of some issues between a FS dependency (APR) and the FBSD threading model. Still, on a light load I wouldn't expect this kind of behavior. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/87d18253/attachment.html From mrene_lists at avgs.ca Fri Jun 12 16:34:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 12 Jun 2009 19:34:52 -0400 Subject: [Freeswitch-users] high latency In-Reply-To: <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> Message-ID: Try Math On 12-Jun-09, at 7:28 PM, Michael Collins wrote: > > > On Fri, Jun 12, 2009 at 3:53 PM, David Burgess > wrote: > On Fri, Jun 12, 2009 at 4:33 PM, Michael Collins > wrote: > > Can you describe your setup? Need to know what kind of OS and > hardware is > > running FS as well as what kind of phones. Any NAT involved? > > -MC > > FS is running inside pfsense, which is a freeBSD-based firewall > (pfsense.org). Hardware is a lightly loaded Soekris net5501 (500 MHz > Geode x86, 512 MB RAM)(www.soekris.com). > > FS external and internal profiles are both listening on WAN with a > public IP address. My extensions are 2 lines on a Linksys PAP2T > residing on the LAN, registered to the internal profile on the WAN > interface. All calls are made via a sip trunk. > > PAP2T normally reports a decode latency in the neighborhood of 30 ms > and jitter of 5 ms. Ping time to my provider's rtp servers is ~42 ms. > > Every call starts out with imperceptible latency, but at some point > the caller notices a long delay, as I said, usually around 3 seconds, > and usually only on calls lasting 500 seconds or more. > > Anything else I can provide? > > Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen > anything like this with FS+pfSense. The FS devs aren't exactly keen > on FS + FBSD because of some issues between a FS dependency (APR) > and the FBSD threading model. Still, on a light load I wouldn't > expect this kind of behavior. > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/1d67d0f2/attachment.html From john at feith.com Fri Jun 12 17:28:18 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 20:28:18 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130028.n5D0SIUF008628@jwlab.FEITH.COM> Upgraded from Apr 3 svn to svn 13769. Calling from openzap to 9999 (music on hold) works. Calling from openzap to 9995 (5 sec echo test) works. Calling from openzap to vmail works. Calling from Grandstream to 9999 (music on hold) works. Calling from Grandstream to 9995 (5 sec echo test) doesn't work ... call goes through however silence is heard. Calling from Grandstream to vmail doesn't work ... call goes through however vmail disconnects apparently due to receiving silence. Calling from Grandstream to openzap doesn't work ... call goes through and the Grandstream can hear what is said on the openzap side, however openzap hears silence from the Grandstream. Calling from Grandstream to Grandstream doesn't work ... call goes through however both sides hear silence. Suggestions on how to proceed? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From john at feith.com Fri Jun 12 17:48:21 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 20:48:21 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130048.n5D0mLAx008649@jwlab.FEITH.COM> BTW: in all cases show channels says PCMU 8000 is being used for the read and well as write codec. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From diego.viola at gmail.com Fri Jun 12 17:50:04 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 20:50:04 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH Message-ID: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> Hi everyone, I just found this project which seems to be a new VM for running languages... the parrot project aims to create a virtual machine for Perl 6 and other dynamic languages. You can take a look at it here: http://www.parrot.org/ It already supports many different languages: http://www.parrot.org/languages And it has a Apache module: http://www.parrot.org/mod_parrot I wonder that makes it embeddable... would it be possible to create a mod_parrot for FreeSWITCH? Please take this only as a feedback and not as anything else. Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/c318ed33/attachment-0001.html From diego.viola at gmail.com Fri Jun 12 17:53:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 20:53:05 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> Message-ID: <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> Here mercutioviz gave me some interesting info about Parrot. 20:47 <@mercutioviz> diegoviola: parrot isn't tied specifically to perl 6 but it is an offshoot of the perl 6 effort 20:47 <@mercutioviz> they were smart to break apart the big project into two separate projects 20:47 <@mercutioviz> parrot is strictly a virtual machine 20:48 <@mercutioviz> perl 6 is strictly a definition of a scripting language 20:48 <@mercutioviz> that lets people port all sorts of stuff to parrot without tripping over perl 6 20:49 <@mercutioviz> it also creates freakish possibilities, like calling perl library functions from a python script Regards, Diego On Fri, Jun 12, 2009 at 8:50 PM, Diego Viola wrote: > Hi everyone, > > I just found this project which seems to be a new VM for running > languages... the parrot project aims to create a virtual machine for Perl 6 > and other dynamic languages. You can take a look at it here: > > http://www.parrot.org/ > > It already supports many different languages: > > http://www.parrot.org/languages > > And it has a Apache module: > > http://www.parrot.org/mod_parrot > > I wonder that makes it embeddable... would it be possible to create a > mod_parrot for FreeSWITCH? > > Please take this only as a feedback and not as anything else. > > Diego > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/dee5f3b9/attachment.html From msc at freeswitch.org Fri Jun 12 17:56:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 17:56:03 -0700 Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn In-Reply-To: <200906130028.n5D0SIUF008628@jwlab.FEITH.COM> References: <200906130028.n5D0SIUF008628@jwlab.FEITH.COM> Message-ID: <87f2f3b90906121756l4421b309k180f677f6a51b4a3@mail.gmail.com> On Fri, Jun 12, 2009 at 5:28 PM, John Wehle wrote: > Upgraded from Apr 3 svn to svn 13769. > That's a pretty decent jump. I think possibly that the configs changed, especially the auto-nat stuff. For kicks, try launching freeswitch with the "-nonat" flag and see if your symptoms persist. It may be that you need to get newer versions of the sip profile config files. If you didn't make any modifications to internal.xml and external.xml then delete those from conf/sip_profiles and then go to your fs source and do a "make samples" to get fresh copies of those two files. If you have modified those two files then I recommend looking at the new default config versions of those two files and integrating your changes into the new ones. Let us know how it goes... -MC > Calling from openzap to 9999 (music on hold) works. > Calling from openzap to 9995 (5 sec echo test) works. > Calling from openzap to vmail works. > > Calling from Grandstream to 9999 (music on hold) works. > Calling from Grandstream to 9995 (5 sec echo test) doesn't work > ... call goes through however silence is heard. > Calling from Grandstream to vmail doesn't work ... call goes > through however vmail disconnects apparently due to receiving silence. > > Calling from Grandstream to openzap doesn't work ... call goes > through and the Grandstream can hear what is said on the openzap > side, however openzap hears silence from the Grandstream. > > Calling from Grandstream to Grandstream doesn't work ... call goes > through however both sides hear silence. > > Suggestions on how to proceed? > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/45d58b19/attachment.html From apt.get at gmail.com Fri Jun 12 17:58:34 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 12 Jun 2009 18:58:34 -0600 Subject: [Freeswitch-users] high latency In-Reply-To: References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> Message-ID: On Fri, Jun 12, 2009 at 5:34 PM, Mathieu Rene wrote: > Try Thanks, I will try that. > Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen anything like > this with FS+pfSense. Yeah, we've discussed it. He's experienced the same thing but doesn't have an answer at this point. The FS devs aren't exactly keen on FS + FBSD because > of some issues between a FS dependency (APR) and the FBSD threading model. > Still, on a light load I wouldn't expect this kind of behavior. Interesting. Thanks for the feedback. I'll let the list know what I find out. db From diego.viola at gmail.com Fri Jun 12 18:09:25 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 21:09:25 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> Message-ID: <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> They seem to have an embedding API here. http://docs.parrot.org/parrot/latest/html/docs/pdds/draft/pdd10_embedding.pod.html Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot support also. Diego On Fri, Jun 12, 2009 at 8:53 PM, Diego Viola wrote: > Here mercutioviz gave me some interesting info about Parrot. > > 20:47 <@mercutioviz> diegoviola: parrot isn't tied specifically to perl 6 > but it is an offshoot of the perl 6 effort > 20:47 <@mercutioviz> they were smart to break apart the big project into > two separate projects > 20:47 <@mercutioviz> parrot is strictly a virtual machine > 20:48 <@mercutioviz> perl 6 is strictly a definition of a scripting > language > 20:48 <@mercutioviz> that lets people port all sorts of stuff to parrot > without tripping over perl 6 > 20:49 <@mercutioviz> it also creates freakish possibilities, like calling > perl library functions from a python script > > Regards, > > Diego > > > On Fri, Jun 12, 2009 at 8:50 PM, Diego Viola wrote: > >> Hi everyone, >> >> I just found this project which seems to be a new VM for running >> languages... the parrot project aims to create a virtual machine for Perl 6 >> and other dynamic languages. You can take a look at it here: >> >> http://www.parrot.org/ >> >> It already supports many different languages: >> >> http://www.parrot.org/languages >> >> And it has a Apache module: >> >> http://www.parrot.org/mod_parrot >> >> I wonder that makes it embeddable... would it be possible to create a >> mod_parrot for FreeSWITCH? >> >> Please take this only as a feedback and not as anything else. >> >> Diego >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/6964502b/attachment.html From jason at jasonjgw.net Fri Jun 12 18:15:07 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 13 Jun 2009 11:15:07 +1000 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> Message-ID: <20090613011507.GA13918@jdc.jasonjgw.net> Diego Viola wrote: > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot > support also. Are you offering to write it? From diego.viola at gmail.com Fri Jun 12 18:17:42 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 21:17:42 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <20090613011507.GA13918@jdc.jasonjgw.net> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> <20090613011507.GA13918@jdc.jasonjgw.net> Message-ID: <86a32abc0906121817n28a9ddcdu1d29b68b16aff66e@mail.gmail.com> No, I'm not a C programmer, just offering the idea (feedback). On Fri, Jun 12, 2009 at 9:15 PM, Jason White wrote: > Diego Viola wrote: > > > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot > > support also. > > Are you offering to write it? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/d8c27371/attachment.html From john at feith.com Fri Jun 12 18:19:22 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 21:19:22 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130119.n5D1JMs0008730@jwlab.FEITH.COM> Yet more information ... a packet trace of a openzap to Grandstream call shows: Source Destination Packet FreeSWITCH Grandstream SIP Request: INVITE ... Grandstream FreeSWITCH SIP Status: 100 Trying Grandstream FreeSWITCH SIP Status: 180 Ringing Grandstream FreeSWITCH SIP Status: 200, with session description FreeSWITCH Grandstream SIP Request: ACK ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCH Grandstream SIP Request: BYE ... Grandstream FreeSWITCH SIP Status: 200 OK The interesting thing is I don't see the Grandstream attempt to send audio. Is there something that FreeSWITCH needs to say to the Grandstream in order for the phone to send audio? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From diego.viola at gmail.com Fri Jun 12 18:19:26 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 21:19:26 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121817n28a9ddcdu1d29b68b16aff66e@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> <20090613011507.GA13918@jdc.jasonjgw.net> <86a32abc0906121817n28a9ddcdu1d29b68b16aff66e@mail.gmail.com> Message-ID: <86a32abc0906121819i6f9b5f27hc505cd145d738979@mail.gmail.com> Could I add a bounty? On Fri, Jun 12, 2009 at 9:17 PM, Diego Viola wrote: > No, I'm not a C programmer, just offering the idea (feedback). > > > On Fri, Jun 12, 2009 at 9:15 PM, Jason White wrote: > >> Diego Viola wrote: >> >> > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot >> > support also. >> >> Are you offering to write it? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/47f8739f/attachment.html From john at feith.com Fri Jun 12 19:24:24 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 22:24:24 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130224.n5D2OOC3008870@jwlab.FEITH.COM> > I think possibly that the configs changed, specially the auto-nat stuff Yep ... a closer look at the packet trace showed FreeSWITCH settings the Contact as 10.10.10.1 instead of the actual IP address of the machine. > If you have modified those two files then I recommend looking at the new > default config versions of those two files and integrating your changes into > the new ones. Yep ... that's my SOP. Looking at the internal.xml supplied with the new FS I see: Once I commented out those entries everything worked fine. I'm kind of surprised that this default changed ... the older FS came with these commented out and worked fine in the simple configuration where the server and phones are on the same network segment. In any case my config has been adjusted, things are working, it's Friday, and I get to go home so life is good. :-) -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From mcampbellsmith at gmail.com Sat Jun 13 01:44:00 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sat, 13 Jun 2009 18:44:00 +1000 Subject: [Freeswitch-users] Remove voicemail prompts Message-ID: <33c87fa30906130144s2ee1fc42n1c20b1ff2acfefb0@mail.gmail.com> Hi! How can I configure voicemail so that I do not get the options such as "record your message at the tone" and "mark this message as urgent" Thanks! From brian at freeswitch.org Sat Jun 13 09:14:40 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 12:14:40 -0400 Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn In-Reply-To: <200906130224.n5D2OOC3008870@jwlab.FEITH.COM> References: <200906130224.n5D2OOC3008870@jwlab.FEITH.COM> Message-ID: The bigger question is why was it finding that IP if it was wrong. /b On Jun 12, 2009, at 10:24 PM, John Wehle wrote: > Yep ... that's my SOP. > > Looking at the internal.xml supplied with the new FS I see: > > > > > Once I commented out those entries everything worked fine. > > I'm kind of surprised that this default changed ... the older FS came > with these commented out and worked fine in the simple configuration > where the server and phones are on the same network segment. > > In any case my config has been adjusted, things are working, it's > Friday, and I get to go home so life is good. :-) > > -- John Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/f1701780/attachment.html From wiltingtree at gmail.com Sat Jun 13 09:18:54 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 13 Jun 2009 12:18:54 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly Message-ID: Hi, I have a question about mod_voicemail. I would like to use it independently of the dialplan and I was hoping to be able to add new voicemail accounts on-the-fly, without having to update the FreeSWITCH configuration files. But it seems to be forcing me to manually add each user to the dialplan. Does anybody know if mod_voicemail was made to work the way I'm trying to use it, and the best way to approach this? Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/d5d5b9e7/attachment.html From Mailings at kh-dev.de Sat Jun 13 09:30:33 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sat, 13 Jun 2009 18:30:33 +0200 Subject: [Freeswitch-users] t38modem without registration... Message-ID: Hi all, maybe someone can help me here. For some compatibility issues I need to use an older version of t38modem which doesn't support SIP registration. Incoming faxes work well, but I don't know how to set this up for outgoing faxes. So, how can I setup this with FS for outgoing faxes without SIP registration? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/3a4fbec5/attachment.html From brian at freeswitch.org Sat Jun 13 09:58:57 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 12:58:57 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: Why would have to touch the dialplan? Just one regex to catch it all.. and send it to voicemail. You still have to add users to the directory for them to have a mailbox anyway. /b On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > Hi, I have a question about mod_voicemail. I would like to use it > independently of the dialplan and I was hoping to be able to add new > voicemail accounts on-the-fly, without having to update the > FreeSWITCH configuration files. But it seems to be forcing me to > manually add each user to the dialplan. > > Does anybody know if mod_voicemail was made to work the way I'm > trying to use it, and the best way to approach this? > > Thanks, > Adam > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment-0001.html From wiltingtree at gmail.com Sat Jun 13 11:11:06 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 13 Jun 2009 14:11:06 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly Message-ID: I'm sorry, I said dialplan, but I was talking about the directory. So the directory file must be edited every time a new mailbox is created? > > Message: 7 > Date: Sat, 13 Jun 2009 12:58:57 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Why would have to touch the dialplan? Just one regex to catch it > all.. and send it to voicemail. You still have to add users to the > directory for them to have a mailbox anyway. > > /b > > On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > > > Hi, I have a question about mod_voicemail. I would like to use it > > independently of the dialplan and I was hoping to be able to add new > > voicemail accounts on-the-fly, without having to update the > > FreeSWITCH configuration files. But it seems to be forcing me to > > manually add each user to the dialplan. > > > > Does anybody know if mod_voicemail was made to work the way I'm > > trying to use it, and the best way to approach this? > > > > Thanks, > > Adam > > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 129 > ************************************************* > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/52ac1b3e/attachment.html From krice at freeswitch.org Sat Jun 13 11:25:32 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 13 Jun 2009 13:25:32 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: Message-ID: Either that or you feed it from something else like XML_CURL From: Adam Wilt Reply-To: Date: Sat, 13 Jun 2009 14:11:06 -0400 To: Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly I'm sorry, I said dialplan, but I was talking about the directory.? So the directory file must be edited every time a new mailbox is created? ? > > Message: 7 > Date: Sat, 13 Jun 2009 12:58:57 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Why would have to touch the dialplan? ?Just one regex to catch it > all.. and send it to voicemail. ?You still have to add users to the > directory for them to have a mailbox anyway. > > /b > > On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > >> > Hi, I have a question about mod_voicemail. I would like to use it >> > independently of the dialplan and I was hoping to be able to add new >> > voicemail accounts on-the-fly, without having to update the >> > FreeSWITCH configuration files. But it seems to be forcing me to >> > manually add each user to the dialplan. >> > >> > Does anybody know if mod_voicemail was made to work the way I'm >> > trying to use it, and the best way to approach this? >> > >> > Thanks, >> > Adam >> > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/ba > d009e9/attachment.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 129 > ************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/7bd28f05/attachment.html From sprice at gmail.com Sat Jun 13 11:31:26 2009 From: sprice at gmail.com (SP) Date: Sat, 13 Jun 2009 13:31:26 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: <7e2ac3270906131131w6ec5c80bvc29e98ca8c252b9c@mail.gmail.com> Unless you use mod_xml_curl On Sat, Jun 13, 2009 at 13:11, Adam Wilt wrote: > I'm sorry, I said dialplan, but I was talking about the directory. > So the directory file must be edited every time a new mailbox is created? > > > >> >> Message: 7 >> Date: Sat, 13 Jun 2009 12:58:57 -0400 >> From: Brian West >> Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="us-ascii" >> >> >> Why would have to touch the dialplan? Just one regex to catch it >> all.. and send it to voicemail. You still have to add users to the >> directory for them to have a mailbox anyway. >> >> /b >> >> On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: >> >> > Hi, I have a question about mod_voicemail. I would like to use it >> > independently of the dialplan and I was hoping to be able to add new >> > voicemail accounts on-the-fly, without having to update the >> > FreeSWITCH configuration files. But it seems to be forcing me to >> > manually add each user to the dialplan. >> > >> > Does anybody know if mod_voicemail was made to work the way I'm >> > trying to use it, and the best way to approach this? >> > >> > Thanks, >> > Adam >> > >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of Freeswitch-users Digest, Vol 36, Issue 129 >> ************************************************* >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/dfb8a0aa/attachment.html From christian.loeschenkohl at xpirio.com Sat Jun 13 10:57:52 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sat, 13 Jun 2009 19:57:52 +0200 Subject: [Freeswitch-users] mod_php needed Message-ID: <4A33E8A0.1070708@xpirio.com> hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian L?schenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From larclap at yahoo.com Sat Jun 13 12:42:15 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 13 Jun 2009 12:42:15 -0700 Subject: [Freeswitch-users] Error in Dialplan documentation? Message-ID: <006e01c9ec5f$0de046a0$29a0d3e0$@com> At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the top under From Dialplan, it says: Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml. However, I cannot find ${sip_profile} in freeswitch.xml. Is the documentation correct? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/b564c8ca/attachment-0001.html From brian at freeswitch.org Sat Jun 13 13:08:21 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 15:08:21 -0500 Subject: [Freeswitch-users] Error in Dialplan documentation? In-Reply-To: <006e01c9ec5f$0de046a0$29a0d3e0$@com> References: <006e01c9ec5f$0de046a0$29a0d3e0$@com> Message-ID: Yes look at the default dialplan... you should note that its in the default only. /b On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote: > At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, > near the top under From Dialplan, it says: > Bridge the incoming call to extension 100 and 101. The '%' is used > instead of the @ to indicate that the endpoints are registered > locally. Separate multiple endpoints with a comma. The $ > {sip_profile} variable is defined in freeswitch.xml. > > However, I cannot find ${sip_profile} in freeswitch.xml. Is the > documentation correct? > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/7690bdbe/attachment.html From msc at freeswitch.org Sat Jun 13 13:56:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 13 Jun 2009 13:56:48 -0700 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <4A33E8A0.1070708@xpirio.com> References: <4A33E8A0.1070708@xpirio.com> Message-ID: <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian L?schenkohl > hello > > i am working for an austrian voip carrier. > for a few months i work with freeswitch and it is simply great. > it solves our needs in many places (high volume, flexible, stable). > the only thing i really miss is the avalibilty of php as a call control > language. > mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't > that good (or even there :-) ). > i know there is perl, i also implemented some applications (conference > system with provisioning, > inbound call routing to our application servers, some tests as pbx), but > what should i say - > perl and me aren't compatible in the end. > > the greatest thing for us would be that mod_php comes alive again with the > functional state > of mod_perl ( > http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > there is also a bounty entry about mod_php, to pay for this would also be > an option and > could be discussed. > > keep on with the excellent work and greetings from austria > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung& Entwicklung VoIP > > xpirio > Telekommunikation& Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/4c438e90/attachment.html From larclap at yahoo.com Sat Jun 13 14:16:13 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 13 Jun 2009 14:16:13 -0700 Subject: [Freeswitch-users] Error in Dialplan documentation? In-Reply-To: References: <006e01c9ec5f$0de046a0$29a0d3e0$@com> Message-ID: <008601c9ec6c$2ea0c100$8be24300$@com> I'm sorry, but I can't find ${sip_profile} defined in any document in the conf directory. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, June 13, 2009 1:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error in Dialplan documentation? Yes look at the default dialplan... you should note that its in the default only. /b On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote: At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the top under From Dialplan, it says: Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml. However, I cannot find ${sip_profile} in freeswitch.xml. Is the documentation correct? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/b68fae43/attachment.html From brian at freeswitch.org Sat Jun 13 14:21:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 16:21:30 -0500 Subject: [Freeswitch-users] Error in Dialplan documentation? In-Reply-To: <008601c9ec6c$2ea0c100$8be24300$@com> References: <006e01c9ec5f$0de046a0$29a0d3e0$@com> <008601c9ec6c$2ea0c100$8be24300$@com> Message-ID: <86F9DFF8-8A2C-49FF-9AD2-771529E9BE23@freeswitch.org> Its not there anymore... use_profile is. But its just a variable so leaving it as is .. its prob. the best.. /b On Jun 13, 2009, at 4:16 PM, Lars Zeb wrote: > I?m sorry, but I can?t find ${sip_profile} defined in any document > in the conf directory. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Saturday, June 13, 2009 1:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error in Dialplan documentation? > > Yes look at the default dialplan... you should note that its in the > default only. > > /b > > On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote: > > > At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, > near the top under From Dialplan, it says: > Bridge the incoming call to extension 100 and 101. The '%' is used > instead of the @ to indicate that the endpoints are registered > locally. Separate multiple endpoints with a comma. The $ > {sip_profile} variable is defined in freeswitch.xml. > > However, I cannot find ${sip_profile} in freeswitch.xml. Is the > documentation correct? > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bbcd6c82/attachment-0001.html From wiltingtree at gmail.com Sat Jun 13 15:28:55 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 13 Jun 2009 18:28:55 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly Message-ID: Thanks. I would really like mod_voicemail to be database driven, instead of by XML and cURL. I noticed in the documentation that you can provide an ODBC handle in the voicemail.conf.xml file, and so I tried it with MySQL. It created the two tables, voicemail_msgs and voicemail_prefs properly, and voicemail_msgs works the way I expect; when a new voicemail is created it writes a new record in this table. But I don't understand the purpose of voicemail_prefs; when I add a record here with a username and password, mod_voicemail ignores it. So I still have to use the config file or xml_curl to set-up the users. It doesn't seem like mod_voicemail reads from or writes to this table. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/83729b49/attachment.html From krice at freeswitch.org Sat Jun 13 15:41:38 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 13 Jun 2009 17:41:38 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: Message-ID: That does not provide for User configuration only voicemail metadata storage... Again, at this time you would need to use XML CURL to do what you are trying to do. There at some point in the future may be a way to grab users via direct ODBC or other database driver however at this point its not possible... Your only other possible option is via LDAP for the users From: Adam Wilt Reply-To: Date: Sat, 13 Jun 2009 18:28:55 -0400 To: Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly Thanks. I?would really like?mod_voicemail to be database driven, instead of by XML and cURL. I noticed in the documentation that you can provide an ODBC handle in the voicemail.conf.xml file, and so I tried it with MySQL. It created the two tables, voicemail_msgs and voicemail_prefs properly, and voicemail_msgs works the way I expect; when a new voicemail is created it writes a new record in this table.? But I don't understand the purpose of voicemail_prefs;? when I add a record here with a username and password, mod_voicemail ignores it. So I still have to use the config file or xml_curl to set-up the users. It doesn't seem like mod_voicemail reads from or writes to this table. ? ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/553eeec1/attachment.html From brian at freeswitch.org Sat Jun 13 15:42:12 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 17:42:12 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: Then you're free to write your own directory hook and plug it directly into the database how ever you wish. Look how XML_CURL uses this interface. /b On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote: > Thanks. I would really like mod_voicemail to be database driven, > instead of by XML and cURL. > I noticed in the documentation that you can provide an ODBC handle > in the voicemail.conf.xml file, and so I tried it with MySQL. > It created the two tables, voicemail_msgs and voicemail_prefs > properly, and voicemail_msgs works the way I expect; when a new > voicemail is created it writes a new record in this table. But I > don't understand the purpose of voicemail_prefs; when I add a > record here with a username and password, mod_voicemail ignores it. > So I still have to use the config file or xml_curl to set-up the > users. It doesn't seem like mod_voicemail reads from or writes to > this table. From diego.viola at gmail.com Sat Jun 13 16:07:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 13 Jun 2009 19:07:02 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: <86a32abc0906131607k6c67dc8ay95ad1f7948498317@mail.gmail.com> And please contribute things back when you do it. Diego On Sat, Jun 13, 2009 at 6:42 PM, Brian West wrote: > Then you're free to write your own directory hook and plug it directly > into the database how ever you wish. Look how XML_CURL uses this > interface. > > /b > > On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote: > > > Thanks. I would really like mod_voicemail to be database driven, > > instead of by XML and cURL. > > I noticed in the documentation that you can provide an ODBC handle > > in the voicemail.conf.xml file, and so I tried it with MySQL. > > It created the two tables, voicemail_msgs and voicemail_prefs > > properly, and voicemail_msgs works the way I expect; when a new > > voicemail is created it writes a new record in this table. But I > > don't understand the purpose of voicemail_prefs; when I add a > > record here with a username and password, mod_voicemail ignores it. > > So I still have to use the config file or xml_curl to set-up the > > users. It doesn't seem like mod_voicemail reads from or writes to > > this table. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/9033e215/attachment.html From nik.middleton at noblesolutions.co.uk Sat Jun 13 16:15:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 14 Jun 2009 00:15:53 +0100 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> Message-ID: I couldn't agree more. We're working with a group that are developing a massive PHP based music application. They are experts in PHP and MySQL but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate with the FS event socket, allows them to work on the areas they know best and not worry about the telephony side too much. We went the lua route, and don't use the dial plan at all. My view is to keep all db access and processing out of FS as much as possible. With the event socket you simply don't need to embed anything apart from the essentials. We are now processing 100,000+ call setups a day (4 hours) per server all using php scripts to drive the application. We may well ultimately use C++ instead of PHP for the event socket comms, but right now PHP does just fine. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 13 June 2009 21:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_php needed Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian L?schenkohl hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian L?schenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090614/2cd4e41f/attachment-0001.html From darklion11 at yahoo.com Sat Jun 13 23:11:43 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sat, 13 Jun 2009 23:11:43 -0700 (PDT) Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? Message-ID: <24018568.post@talk.nabble.com> Ip Windows: 192.168.0.104 Ip Linux: 192.168.0.105 My windows: My sample on sip_profiles/external/dialus.xml My linux: My sample on sip_profiles/external/dialus2.xml I have a number on windows 01497710001, on linux 01497710002 Trying to call each other on windows I dial 0149771002 But error on switch_ivr_originate: INVALID_NUMBER_FORMAT Please help me with this urgent issue... Or send me instructions or xml code that will me to solve this issue... Thanks, Edmar -- View this message in context: http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sun Jun 14 09:39:15 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 14 Jun 2009 11:39:15 -0500 Subject: [Freeswitch-users] MPL Confusion In-Reply-To: References: Message-ID: <950BE21A-5B01-4983-8AF3-1C327EEC5B7D@freeswitch.org> For clarification ... Read section 3.2 and 3.3 of the MPL 1.1 The simplest way I can describe it is how it was described to me "What's yours is yours and what's mine is mine!". /b On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: > I have some confusion about FreeSWITCH's Mozilla Public License 1.1. > I do understand that me or any one can change provided code > according to our customization needs and we are not bound to share > our changes as long as we are not distributing it, right? From gavin.henry at gmail.com Sun Jun 14 13:14:48 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Sun, 14 Jun 2009 21:14:48 +0100 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: <13ca621c0906141314sc9abde9p958fec8cb4772491@mail.gmail.com> Are there any ldap examples? On 13/06/2009, Ken Rice wrote: > That does not provide for User configuration only voicemail metadata > storage... Again, at this time you would need to use XML CURL to do what you > are trying to do. There at some point in the future may be a way to grab > users via direct ODBC or other database driver however at this point its not > possible... Your only other possible option is via LDAP for the users > > > > From: Adam Wilt > Reply-To: > Date: Sat, 13 Jun 2009 18:28:55 -0400 > To: > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > > Thanks. I?would really like?mod_voicemail to be database driven, instead of > by XML and cURL. > I noticed in the documentation that you can provide an ODBC handle in the > voicemail.conf.xml file, and so I tried it with MySQL. > It created the two tables, voicemail_msgs and voicemail_prefs properly, and > voicemail_msgs works the way I expect; when a new voicemail is created it > writes a new record in this table.? But I don't understand the purpose of > voicemail_prefs;? when I add a record here with a username and password, > mod_voicemail ignores it. So I still have to use the config file or xml_curl > to set-up the users. It doesn't seem like mod_voicemail reads from or writes > to this table. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From gavin.henry at gmail.com Sun Jun 14 13:34:15 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Sun, 14 Jun 2009 21:34:15 +0100 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? Message-ID: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From nik.middleton at noblesolutions.co.uk Sun Jun 14 14:38:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 14 Jun 2009 22:38:02 +0100 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? In-Reply-To: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> References: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> Message-ID: Anything that's dedicated undoubtedly has less load that something that's multifunctioned. However the lack of any conversations on front ending a SIP server to FS would likely indicate that no one's found a requirement for it at this time. I would truly hate to see discussions of theoretical performance advantages of one SIP server over another, when in my view, I have yet to reach any real world limit with FS. My FS servers are handling 100,000+ calls/day per server and are probably only at 50% capacity. (I see no point in beating a server to pulp when it's relatively cheap to add another if required) Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gavin Henry Sent: 14 June 2009 21:34 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From drago at windstream.net Sun Jun 14 16:11:27 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 14 Jun 2009 19:11:27 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Message-ID: <000001c9ed45$72058320$56108960$@net> Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: "MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The 'From' field's address to be in the format ''; otherwise, MS Exchange drops the call." I don't know if this is the only problem. However, I see exactly this behavior: "PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established." After "302 (Moved Temporarily", FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090614/ecf43fe4/attachment.html From jingwei.yang at gmail.com Sun Jun 14 19:29:24 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 15 Jun 2009 10:29:24 +0800 Subject: [Freeswitch-users] Possible to initiate multiple skype interfaces with the same id? Message-ID: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> Hi Team, As the subject indicates, is there a possible way to do that? I've tried setting up two different skype instances with the same id in /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh /usr/bin/Xvfb :101 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype --pipelogin &" /usr/bin/Xvfb :102 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype --pipelogin &" The script can be run without any error and when I executed the ps command, I noticed there's only one skype process up. This is strange since normally there will be two (with two different ids) 3782 pts/1 00:00:00 Xvfb 3793 pts/1 00:00:00 dbus-launch 3800 pts/1 00:00:04 Xvfb 3805 pts/1 00:00:03 *skype* 3811 pts/1 00:00:00 dbus-launch Then I started freeswitch and hit an error when loading mod_skypiax: *freeswitch at localhost.localdomain> load mod_skypiax 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING interface_id=1 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error code 3 from X Server 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending message failed with status 3 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error code 3 from X Server 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending message failed with status 3 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==userAAA API CALL [load(mod_skypiax)] output: -ERR [module load file routine returned an error] 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' has been found. Please (re)launch a Skype client logged in as 'userAAA'. Skypiax exiting now 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_skypiax.so **Module load routine returned an error** * Please enlighten whether it's possible to start multiple skype instances with the same skype id. If possible, what are the correct configs? Thanks, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/c438c7cb/attachment-0001.html From msc at freeswitch.org Sun Jun 14 19:51:38 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 14 Jun 2009 19:51:38 -0700 Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? In-Reply-To: <24018568.post@talk.nabble.com> References: <24018568.post@talk.nabble.com> Message-ID: <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> Just curious, why are you dialing out the external gw? -MC Sent from my iPhone On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote: > > Ip Windows: 192.168.0.104 > Ip Linux: 192.168.0.105 > > > My windows: > > My sample on sip_profiles/external/dialus.xml > > > > > > data="sofia/external/$1 at 192.168.0.105:5080"/> > > > > > > My linux: > > My sample on sip_profiles/external/dialus2.xml > > > > > > data="sofia/external/$1 at 192.168.0.104:5080"/> > > > > > > I have a number on windows 01497710001, on linux 01497710002 > > Trying to call each other on windows I dial 0149771002 > > But error on switch_ivr_originate: INVALID_NUMBER_FORMAT > > Please help me with this urgent issue... > > Or send me instructions or xml code that will me to solve this > issue... > > > Thanks, > > Edmar > -- > View this message in context: http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul.degt at gmail.com Sun Jun 14 20:59:38 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Sun, 14 Jun 2009 23:59:38 -0400 Subject: [Freeswitch-users] session.getDigits() not working Message-ID: <4A35C72A.6030804@gmail.com> Trying out latest trunk ans seems like js function session.getDigits() stopped working (not collecting any digits), I do see switch_rtp.c:1560 Send end packet for [5] ts=2222260 dur=2080/2080/2000 seq=8732 in debug log so I assume dtmf is ok. Anybody can shed some light on why wouldn't it work now? Works just fine under 1.0.3 release. I use slightly modified version of disa.js from fs examples. Thanks. From shaheryarkh at googlemail.com Sun Jun 14 21:07:23 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 15 Jun 2009 10:07:23 +0600 Subject: [Freeswitch-users] MPL Confusion In-Reply-To: <950BE21A-5B01-4983-8AF3-1C327EEC5B7D@freeswitch.org> References: <950BE21A-5B01-4983-8AF3-1C327EEC5B7D@freeswitch.org> Message-ID: Thanks, I will look at it in more details as you suggested. I try to be online to discuss mod_msn and mod_yahoo on FS IRC channel this after noon Danish time. Thank you. On Sun, Jun 14, 2009 at 10:39 PM, Brian West wrote: > For clarification ... Read section 3.2 and 3.3 of the MPL 1.1 > > The simplest way I can describe it is how it was described to me > "What's yours is yours and what's mine is mine!". > > /b > > On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: > > > I have some confusion about FreeSWITCH's Mozilla Public License 1.1. > > I do understand that me or any one can change provided code > > according to our customization needs and we are not bound to share > > our changes as long as we are not distributing it, right? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/260babd8/attachment.html From shaheryarkh at googlemail.com Sun Jun 14 21:21:50 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 15 Jun 2009 10:21:50 +0600 Subject: [Freeswitch-users] Possible to initiate multiple skype interfaces with the same id? In-Reply-To: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> References: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> Message-ID: I don't think you can start more then one Skype instance of same Skype ID from a single machine. You can either login with same Skype ID on two (or more different machines) or use two (or more) different Skype IDs on same machine. Thank you. On Mon, Jun 15, 2009 at 8:29 AM, Jingwei Yang wrote: > Hi Team, > > As the subject indicates, is there a possible way to do that? > > I've tried setting up two different skype instances with the same id in > /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh > > /usr/bin/Xvfb :101 -auth > /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & > su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype > --pipelogin &" > > /usr/bin/Xvfb :102 -auth > /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & > su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype > --pipelogin &" > > The script can be run without any error and when I executed the ps command, > I noticed there's only one skype process up. This is strange since normally > there will be two (with two different ids) > > 3782 pts/1 00:00:00 Xvfb > 3793 pts/1 00:00:00 dbus-launch > 3800 pts/1 00:00:04 Xvfb > 3805 pts/1 00:00:03 *skype* > 3811 pts/1 00:00:00 dbus-launch > > Then I started freeswitch and hit an error when loading mod_skypiax: > > *freeswitch at localhost.localdomain> load mod_skypiax > 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev > 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING > interface_id=1 > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev > 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error > code 3 from X Server > > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() > rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending > message failed with status 3 > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev > 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error > code 3 from X Server > > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() > rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending > message failed with status 3 > 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev > 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 > seconds to find a running Skype client and connect to its SKYPE API for > interface_id=1 > 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev > 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running > Skype client, connected to its SKYPE API for interface_id=1, waiting 60 > seconds for CURRENTUSERHANDLE==userAAA > API CALL [load(mod_skypiax)] output: > -ERR [module load file routine returned an error] > > 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev > 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype client > to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, > interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' has > been found. Please (re)launch a Skype client logged in as 'userAAA'. > Skypiax exiting now > 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_skypiax.so > **Module load routine returned an error** > > * > Please enlighten whether it's possible to start multiple skype instances > with the same skype id. If possible, what are the correct configs? > > Thanks, > -Jingwei > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/b1651808/attachment.html From jingwei.yang at gmail.com Sun Jun 14 22:12:02 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 15 Jun 2009 13:12:02 +0800 Subject: [Freeswitch-users] Possible to initiate multiple skype interfaces with the same id? In-Reply-To: References: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> Message-ID: <13529f9d0906142212m690d4176xa735f54886ee63e8@mail.gmail.com> Hi Muhammad, thanks for the reply. On Mon, Jun 15, 2009 at 12:21 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I don't think you can start more then one Skype instance of same Skype ID > from a single machine. You can either login with same Skype ID on two (or > more different machines) or use two (or more) different Skype IDs on same > machine. > > Thank you. > > > On Mon, Jun 15, 2009 at 8:29 AM, Jingwei Yang wrote: > >> Hi Team, >> >> As the subject indicates, is there a possible way to do that? >> >> I've tried setting up two different skype instances with the same id in >> /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh >> >> /usr/bin/Xvfb :101 -auth >> /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & >> su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype >> --pipelogin &" >> >> /usr/bin/Xvfb :102 -auth >> /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & >> su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype >> --pipelogin &" >> >> The script can be run without any error and when I executed the ps >> command, I noticed there's only one skype process up. This is strange since >> normally there will be two (with two different ids) >> >> 3782 pts/1 00:00:00 Xvfb >> 3793 pts/1 00:00:00 dbus-launch >> 3800 pts/1 00:00:04 Xvfb >> 3805 pts/1 00:00:03 *skype* >> 3811 pts/1 00:00:00 dbus-launch >> >> Then I started freeswitch and hit an error when loading mod_skypiax: >> >> *freeswitch at localhost.localdomain> load mod_skypiax >> 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev >> 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING >> interface_id=1 >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev >> 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error >> code 3 from X Server >> >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() >> rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending >> message failed with status 3 >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev >> 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error >> code 3 from X Server >> >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() >> rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending >> message failed with status 3 >> 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev >> 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 >> seconds to find a running Skype client and connect to its SKYPE API for >> interface_id=1 >> 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev >> 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >> seconds for CURRENTUSERHANDLE==userAAA >> API CALL [load(mod_skypiax)] output: >> -ERR [module load file routine returned an error] >> >> 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev >> 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype >> client to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, >> interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' >> has been found. Please (re)launch a Skype client logged in as 'userAAA'. >> Skypiax exiting now >> 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 >> switch_loadable_module_load_file() Error Loading module >> /usr/local/freeswitch/mod/mod_skypiax.so >> **Module load routine returned an error** >> >> * >> Please enlighten whether it's possible to start multiple skype instances >> with the same skype id. If possible, what are the correct configs? >> >> Thanks, >> -Jingwei >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/3312ad0c/attachment-0001.html From darklion11 at yahoo.com Sun Jun 14 22:24:05 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 14 Jun 2009 22:24:05 -0700 (PDT) Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? In-Reply-To: <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> References: <24018568.post@talk.nabble.com> <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> Message-ID: <1245043445817-3078668.post@n2.nabble.com> Actually, I dont know if my xml code is correct.. Please give me an example of an external profile connecting this 2 OS? And instructions Thanks mercutioviz wrote: > > Just curious, why are you dialing out the external gw? > -MC > > Sent from my iPhone > > On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote: > >> >> Ip Windows: 192.168.0.104 >> Ip Linux: 192.168.0.105 >> >> >> My windows: >> >> My sample on sip_profiles/external/dialus.xml >> >> >> >> >> >> data="sofia/external/$1 at 192.168.0.105:5080"/> >> >> >> >> >> >> My linux: >> >> My sample on sip_profiles/external/dialus2.xml >> >> >> >> >> >> data="sofia/external/$1 at 192.168.0.104:5080"/> >> >> >> >> >> >> I have a number on windows 01497710001, on linux 01497710002 >> >> Trying to call each other on windows I dial 0149771002 >> >> But error on switch_ivr_originate: INVALID_NUMBER_FORMAT >> >> Please help me with this urgent issue... >> >> Or send me instructions or xml code that will me to solve this >> issue... >> >> >> Thanks, >> >> Edmar >> -- >> View this message in context: >> http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp3074838p3078668.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090614/366e2f16/attachment.html From darklion11 at yahoo.com Sun Jun 14 22:27:27 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 14 Jun 2009 22:27:27 -0700 (PDT) Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? In-Reply-To: <1245043445817-3078668.post@n2.nabble.com> References: <24018568.post@talk.nabble.com> <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> <1245043445817-3078668.post@n2.nabble.com> Message-ID: <1245043647184-3078676.post@n2.nabble.com> I just want to connect to this to OS with an external xml code. How can I do that? I get some examples on the wiki but not working. Edmar Cruz wrote: > > Actually, I dont know if my xml code is correct.. Please give me an > example of an external profile connecting this 2 OS? And instructions > > Thanks > > > > mercutioviz wrote: >> >> Just curious, why are you dialing out the external gw? >> -MC >> >> Sent from my iPhone >> >> On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote: >> >>> >>> Ip Windows: 192.168.0.104 >>> Ip Linux: 192.168.0.105 >>> >>> >>> My windows: >>> >>> My sample on sip_profiles/external/dialus.xml >>> >>> >>> >>> >>> >>> >>> data="sofia/external/$1 at 192.168.0.105:5080"/> >>> >>> >>> >>> >>> >>> My linux: >>> >>> My sample on sip_profiles/external/dialus2.xml >>> >>> >>> >>> >>> >>> >>> data="sofia/external/$1 at 192.168.0.104:5080"/> >>> >>> >>> >>> >>> >>> I have a number on windows 01497710001, on linux 01497710002 >>> >>> Trying to call each other on windows I dial 0149771002 >>> >>> But error on switch_ivr_originate: INVALID_NUMBER_FORMAT >>> >>> Please help me with this urgent issue... >>> >>> Or send me instructions or xml code that will me to solve this >>> issue... >>> >>> >>> Thanks, >>> >>> Edmar >>> -- >>> View this message in context: >>> http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp3074838p3078676.html Sent from the freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Sun Jun 14 23:57:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 15 Jun 2009 08:57:52 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A328E8E.6030607@freeswitch.org> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> Message-ID: <4A35F0F0.50406@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Stefan, On 12.06.2009 19:21, Stefan Knoblich wrote: > Umm, you've been doing duplicate work then. :( Well, I implemented just one timer by now. So not much time has been wasted ... > > The version of ozmod_isdn i have been working on is completely stateful and > has a couple of timers already implemented. Very good :) > > And i remember giving you the location of the git repository on IRC, > earlier this year. (But never got any feedback) Sorry, at that time we talked about q931 to pcap. By now I thought state timers are still not done, so there wasn't a reason to test it regarding state timers .... buuuut today things changed and I will download your openzap. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKNfDw4tZeNddg3dwRAkThAJ4iPiZ4ZOAKKWpmKdCbjM8oM5mH6QCeMo5m pV4Y1/hpO7osV8cuInYJd2o= =zoZu -----END PGP SIGNATURE----- From dujinfang at gmail.com Mon Jun 15 00:41:49 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 15:41:49 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls Message-ID: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1a55a71a/attachment.html From Claudio.Cavalera at italtel.it Mon Jun 15 01:30:16 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 15 Jun 2009 10:30:16 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <74170625-2AA0-4FF6-969D-3C850DCD7CA0@jerris.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: >> OTOH there will be a bit of trouble getting the internal state out >> of all those modules and libraries... in particular sofia :D > > We have talked quite some about this, its a major job, easily months > of work for multiple programmers. We would love to do it but > its not > on any roadmaps at this time. > Could this be also achieved in hardware via ATCA ? en.wikipedia.org/wiki/Advanced_Telecommunications_Computing_Architecture Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From gavin.henry at gmail.com Mon Jun 15 01:47:54 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 15 Jun 2009 09:47:54 +0100 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? In-Reply-To: References: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> Message-ID: <13ca621c0906150147t504a31feof75a37d8f478a1f3@mail.gmail.com> OK, thanks. 2009/6/14 Nik Middleton : > Anything that's dedicated undoubtedly has less load that something > that's multifunctioned. ?However the lack of any conversations on front > ending a SIP server to FS would likely indicate that no one's found a > requirement for it at this time. > > I would truly hate to see discussions of theoretical performance > advantages of one SIP server over another, when in my view, I have yet > to reach any real world limit with FS. ?My FS servers are handling > 100,000+ calls/day per server and are probably only at 50% capacity. (I > see no point in beating a server to pulp when it's relatively cheap to > add another if required) > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Gavin Henry > Sent: 14 June 2009 21:34 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of > FS? > > Hi, > > I'm excited reading all the threads about how FS blows Asterisk away > so that you don't need ?OpenSIPS/Kamailio in front of FS. Surely there > must be a point when it would be advisable to do that though, as > mod_sofia can't be as good as a dedicated SIP proxy? > > Thanks. > > -- > Sent from my mobile device > > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From grevenx at me.com Mon Jun 15 02:01:27 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Mon, 15 Jun 2009 11:01:27 +0200 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? In-Reply-To: <13ca621c0906150147t504a31feof75a37d8f478a1f3@mail.gmail.com> References: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> <13ca621c0906150147t504a31feof75a37d8f478a1f3@mail.gmail.com> Message-ID: Well, we currently have a scenario where this seems to be the most logical setup currently. We provide PBX as a Service (SaaS), and want to have a cluster of FreeSwitch servers handling registration and presence. Introducing OpenSIPS in front will allow a couple of features, which I don't see how would be implemented in a "good" way without anything in front of FS: - "true" loadbalancing with the loadbalancer module - Live migration of calls to another server to take FS down for maintenance - no need for 100% SRV support in the SIP clients Best regards, Even Andr? On 15. juni. 2009, at 10.47, Gavin Henry wrote: > OK, thanks. > > > 2009/6/14 Nik Middleton : >> Anything that's dedicated undoubtedly has less load that something >> that's multifunctioned. However the lack of any conversations on >> front >> ending a SIP server to FS would likely indicate that no one's found a >> requirement for it at this time. >> >> I would truly hate to see discussions of theoretical performance >> advantages of one SIP server over another, when in my view, I have >> yet >> to reach any real world limit with FS. My FS servers are handling >> 100,000+ calls/day per server and are probably only at 50% >> capacity. (I >> see no point in beating a server to pulp when it's relatively cheap >> to >> add another if required) >> >> Regards >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Gavin Henry >> Sent: 14 June 2009 21:34 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of >> FS? >> >> Hi, >> >> I'm excited reading all the threads about how FS blows Asterisk away >> so that you don't need OpenSIPS/Kamailio in front of FS. Surely >> there >> must be a point when it would be advisable to do that though, as >> mod_sofia can't be as good as a dedicated SIP proxy? >> >> Thanks. >> >> -- >> Sent from my mobile device >> >> http://www.suretecsystems.com/services/openldap/ >> http://www.suretectelecom.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Mon Jun 15 02:33:07 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 15 Jun 2009 12:33:07 +0300 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Message-ID: <1245058387.4694.5.camel@dk-d820> On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote: > What is the current status of Freeswitch? Can I safely use it in a > large scale commercial environment? How active is the Freeswitch > developer community? Hi Paul - We've used FS over the last 18 months or so to handle millions of calls - some wholesale in/out, some IVR, some calling card, some callthrough - with a total value in the millions of dollars; we have no complaints. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From jingwei.yang at gmail.com Mon Jun 15 02:40:08 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 15 Jun 2009 17:40:08 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session Message-ID: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: And here's how I trigger it: *freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA 2909/userBBB* The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: *freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH->2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) * Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. Thanks! -Jingwei ** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/4b6d83ec/attachment.html From christian.loeschenkohl at xpirio.com Mon Jun 15 02:40:24 2009 From: christian.loeschenkohl at xpirio.com (=?windows-1252?Q?Christian_L=F6schenkohl?=) Date: Mon, 15 Jun 2009 11:40:24 +0200 Subject: [Freeswitch-users] mod_php needed In-Reply-To: References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> Message-ID: <4A361708.8020808@xpirio.com> hi thank you very much for your input i can say for me that i realy tried hard to use the event socket library, but untill now i can't use it like i used all the agi scripts or even mod_perl now. what i do most - in examples, if the server get's an incomming call - find the right user for the number (not that easy because of did in austria), from database or file - build the right dial string for the bridge application (here i miss all the php string functions most) - unsing mod_php functions like setVariable, getVariable, answer, transfer, sleep (i don't see how to do this with the php esl) - or i check if the number is part of a conferencing product and build the right conference setup i think this would also be possible with lua and luasql, but i developed years with phpagi und i'm very used to php in every kind of scripting or how-to-get-a-solution situation (since over 10 years now). for me in our setup it's also the highest goal to get the servers mostly independent of each other. i think nobody of our costumers should be unreachable because a central scripting/event server or also database server has gone away (as developers this happens more often as we would like it to :-)) do not get me wrong, freeswitch is very powerfull and in the near future it will replace nearly all of our asterisk servers. in combination with php the freeswitch plattform would be heaven for me i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php), in combination of the mod_python rewrite (page was last modified in june 2007). br On 2009-06-14 01:15, Nik Middleton wrote: > I couldn?t agree more. We?re working with a group that are developing a > massive PHP based music application. They are experts in PHP and MySQL > but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP > to communicate with the FS event socket, allows them to work on the > areas they know best and not worry about the telephony side too much. > We went the lua route, and don?t use the dial plan at all. My view is > to keep all db access and processing out of FS as much as possible. With > the event socket you simply don?t need to embed anything apart from the > essentials. > > We are now processing 100,000+ call setups a day (4 hours) per server > all using php scripts to drive the application. We may well ultimately > use C++ instead of PHP for the event socket comms, but right now PHP > does just fine. > > Regards > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Michael Collins > *Sent:* 13 June 2009 21:57 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_php needed > > Perhaps you should look at controlling calls via the FreeSWITCH event > socket instead of from the dialplan. The nice thing about the event > socket is that your call control can happen on a separate machine. There > is a PHP module for the ESL (event socket library) and it would be > relatively easy for you to get going. Here are some links to get you > started: > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > http://wiki.freeswitch.org/wiki/Event_Socket > > If you absolutely MUST have call control with scripts inside of the > dialplan then there simply is no better choice than Lua. You can learn > Lua in a few hours, but getting mod_php finished and debugged will take > time, money, and other resources that no one seems willing to spend. > Here is some information to consider: > > http://wiki.freeswitch.org/wiki/Mod_lua > > Come join us on IRC (#freeswitch on irc.freenode.net > ) if you want to discuss this further. > > -MC (IRC: mercutioviz) > > 2009/6/13 Christian L?schenkohl > > > hello > > i am working for an austrian voip carrier. > for a few months i work with freeswitch and it is simply great. > it solves our needs in many places (high volume, flexible, stable). > the only thing i really miss is the avalibilty of php as a call control > language. > mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't > that good (or even there :-) ). > i know there is perl, i also implemented some applications (conference > system with provisioning, > inbound call routing to our application servers, some tests as pbx), but > what should i say - > perl and me aren't compatible in the end. > > the greatest thing for us would be that mod_php comes alive again with > the functional state > of mod_perl > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > there is also a bounty entry about mod_php, to pay for this would also > be an option and > could be discussed. > > keep on with the excellent work and greetings from austria > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung& Entwicklung VoIP > > xpirio > Telekommunikation& Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From darklion11 at yahoo.com Mon Jun 15 02:47:15 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 02:47:15 -0700 (PDT) Subject: [Freeswitch-users] Asterisks to Freeswitch CALL REJECTED Message-ID: <24031735.post@talk.nabble.com> I am trying to call Freeswitch using Asterisks and using a softphone X-Lite but the issue is call rejected by freeswitch? Is their any configuration files to allow asterisks to call to freeswitch? -- View this message in context: http://www.nabble.com/Asterisks-to-Freeswitch-CALL-REJECTED-tp24031735p24031735.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 02:49:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 02:49:04 -0700 (PDT) Subject: [Freeswitch-users] Asterisks to Freeswitch CALL REJECTED Message-ID: <24031735.post@talk.nabble.com> I am trying to call Freeswitch using Asterisks and using a softphone X-Lite but the issue is call rejected by freeswitch? Is their any configuration files to allow asterisks to call to freeswitch? Separate OS am using Linux for Asterisks Windows for freeswitch... -- View this message in context: http://www.nabble.com/Asterisks-to-Freeswitch-CALL-REJECTED-tp24031735p24031735.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 02:59:28 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 02:59:28 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? Message-ID: <24031900.post@talk.nabble.com> is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Mon Jun 15 03:06:32 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 18:06:32 +0800 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24031900.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> Message-ID: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: > > is there any available gui for freeswitch using cake php complete > instead of > wikipbx, spice softphone or pfsense? > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From darklion11 at yahoo.com Mon Jun 15 03:21:45 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 03:21:45 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: References: <24031900.post@talk.nabble.com> Message-ID: <24032171.post@talk.nabble.com> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: > > http://www.tcapi.org/index.php?title=Main_Page > > > > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: > >> >> is there any available gui for freeswitch using cake php complete >> instead of >> wikipbx, spice softphone or pfsense? >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From durk.debeer at isp.solcon.nl Mon Jun 15 03:26:17 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Mon, 15 Jun 2009 12:26:17 +0200 Subject: [Freeswitch-users] funny effect after minimizing xml files Message-ID: Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect /d -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/f7f764a6/attachment.html From dujinfang at gmail.com Mon Jun 15 03:31:20 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 15 Jun 2009 18:31:20 +0800 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24032171.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> Message-ID: <7E486E4D-B08F-4BB3-A1D2-FD97DBACF204@gmail.com> On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote: > > Yup tcapi is a great cake php GUI for freeswitch but it is not yet > fully > developed... > Is there any GUI with billing options? > > AFAIK, no fully developed GUI available yet, just curious, why are you finding a GUI instead of wikipbx or pfsense? > seven-8 wrote: >> >> http://www.tcapi.org/index.php?title=Main_Page >> >> >> >> On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >> >>> >>> is there any available gui for freeswitch using cake php complete >>> instead of >>> wikipbx, spice softphone or pfsense? >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Mon Jun 15 03:35:02 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 15 Jun 2009 20:35:02 +1000 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Message-ID: <20090615103502.GA32625@jdc.jasonjgw.net> Paul Mahler wrote: > I have a large project coming up. I'm interested in using Freeswitch > instead of SER and Asterisk. > > What is the current status of Freeswitch? Can I safely use it in a > large scale commercial environment? How active is the Freeswitch > developer community? Others have already addressed most of your questions. I would like to point out, however, that the FreeSWITCH developers offer support, by way of a consulting service, on a commercial basis. If you're running FreeSWITCH in a commercial setting and encounter complex issues that require expert advice or attention from developers, consider entering into a consulting contract. This will also help to fund the project and ensure that the development community remains as active as we all want it to be. From jason at jasonjgw.net Mon Jun 15 03:38:08 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 15 Jun 2009 20:38:08 +1000 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: <200906151030.n5FATaG5015367@opera.rednote.net> References: <200906151030.n5FATaG5015367@opera.rednote.net> Message-ID: <20090615103808.GB32625@jdc.jasonjgw.net> Durk de Beer wrote: > > Hello I've minimized de xml files where possible to make a dialplan that is > as short as possible. Now do I've this funny effect to dial my extensions > who are running from 200 to 207. It seams that I'm able to dial an > extension in closed in a number. So for instants if I dial 120275 extension > 202 will ring even tried it whit two extensions in a number like 202205 . > This results in the first extension ringing so 202205, 202 will ring > 205202, 205 will ring. At this time I'm unable to pinpoint the cause of > this behaviour. Could someone point me to the cause of this effect I don't understand the problem, but my general advice is this: learn to read the FreeSWITCH logs carefully. Make sure that the log level is set to "debug", as it is in the default configuration, then carefully check the log files to see which dialplan extension matched and how the call was processed. From dujinfang at gmail.com Mon Jun 15 03:46:18 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 15 Jun 2009 18:46:18 +0800 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: <4a36232b.8b13f30a.3877.3cafSMTPIN_ADDED@mx.google.com> References: <4a36232b.8b13f30a.3877.3cafSMTPIN_ADDED@mx.google.com> Message-ID: how to help without seeing your dialplan? On Jun 15, 2009, at 6:26 PM, Durk de Beer wrote: > Hello I've minimized de xml files where possible to make a dialplan > that is as short as possible. Now do I've this funny effect to dial > my extensions who are running from 200 to 207. It seams that I'm > able to dial an extension in closed in a number. So for instants if > I dial 120275 extension 202 will ring even tried it whit two > extensions in a number like 202205 . This results in the first > extension ringing so 202205, 202 will ring 205202, 205 will ring. At > this time I'm unable to pinpoint the cause of this behaviour. Could > someone point me to the cause of this effect > > /d > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/48d87c9c/attachment.html From dave at 3c.co.uk Mon Jun 15 03:53:20 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 15 Jun 2009 13:53:20 +0300 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: <20090615103122.9F69D20169@smtp.3c.co.uk> References: <20090615103122.9F69D20169@smtp.3c.co.uk> Message-ID: <1245063200.4694.7.camel@dk-d820> You've probably deleted the start/end markers from your dialplan matches..? It might be easier to help if you posted or pastebinned your dialplan. --Dave > Hello I've minimized de xml files where possible to make a dialplan > that is as short as possible. Now do I've this funny effect to dial my > extensions who are running from 200 to 207. It seams that I'm able to > dial an extension in closed in a number. So for instants if I dial > 120275 extension 202 will ring even tried it whit two extensions in a > number like 202205 . This results in the first extension ringing so > 202205, 202 will ring 205202, 205 will ring. At this time I'm unable > to pinpoint the cause of this behaviour. Could someone point me to the > cause of this effect > > > /d > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From saeedahmad1981 at gmail.com Mon Jun 15 04:59:50 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Mon, 15 Jun 2009 13:59:50 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> Message-ID: Michael: We are using 5.0, and i think we tested this feature quiet a while ago and there was no CDR problem. Raymond: Thanks for hint i'll try it... On Fri, Jun 12, 2009 at 6:54 PM, Michael Giagnocavo wrote: > Well, Nextone for instance has a database the keeps most of the state of > calls, and it?s replicated between the two nodes. (I seem to recall the > database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, > the CDRs still get truncated when there?s any kind of switchover. > > > > But Nextone is a closed system with limited services. As MikeJ mentioned, > it was discussed for FS, but it?s a LOT of work to get that state > synchronized. And, every custom app/module would have to register and > support recreating their state. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Saeed Ahmed > *Sent:* Friday, June 12, 2009 7:39 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques > > > > No idea at all, > > It?s a commercial SBC. > > I wish if we can have same functionality in FS. > > > > - Saeed > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Even Andr? > Fiskvik > *Sent:* Friday, June 12, 2009 3:04 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques > > > > Can you comment some more on how this is configured? > > Would it be something that could be added to the wiki in the SBC setup > page? > > > > Best regards, > > Even Andr? Fiskvik > > > > On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: > > > > I've experience with a commercial SBC, these are two machines running in > cluster mode. In that case if one SBC is going down then other will take all > new calls including the call which were active on broken SBC (SIP only). > > Thats quite ideal for wholesale traffic where the SBC will never be idle. > > On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh wrote: > > On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > > >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > >>> > >>> Well, if you're running multiple machines, waiting for it to drainstop > >>> isn't that big of a deal unless you're in some sort of hurry, right? > >>> Give it an hour or so to drainstop, then kill 'em. > >> > >> Yes that's exactly what I'm trying to do. The problem is some people > will > >> only try one IP address. > > > > Clients that don't properly implement SRV/NAPTR and fail over need to be > > smacked. :) (not customers but software that fails to do that) > > Yes I'm sure much of their software can do this but it has been set up for > static numeric IPs. And getting the IP changed is a week-long process for > some customers! > > > >>> Would it not be simpler to try to do something with re-invites or > REFER, > >>> assuming your endpoints support it? > >> > >> That was actually plan A. I already added a property in sip_profile > called > >> failover_redirect, which specifies another server to try if FS can't > >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), > >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max > >> Calls In Progress. > > > > You can't send a 302 to a call thats already established. > > Yes and I don't want to touch established calls - those calls can stay > there until they drop. This is sent to new requests when > switch_core_session_request fails in mod_sofia. > > > >> Turns out not all my endpoints support it :( > > > > AKA broken endpoints. :) > > Some are broken. Some just have this feature disabled. For 'security > reasons'. You know the drill. > > > {P^/ > John > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e08ebdf7/attachment-0001.html From gmaruzz at celliax.org Mon Jun 15 05:16:55 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 15 Jun 2009 14:16:55 +0200 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> Message-ID: <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> Hi Jingwel, thanks for reporting. Could you please add a Jira issue with as much details as possible? general guide for reporting bugs: http://wiki.freeswitch.org/wiki/Reporting_Bugs what to add for skypiax: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests mod_skypiax Jira: http://jira.freeswitch.org/browse/MODSKYPIAX Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang wrote: > Hi Team, > > I've been using the record_session feature to record call sessions. Here's > how I prepared the dialplan: > > ??? > ????? > ??????? > ??????? > ????? > ??? > > And here's how I trigger it: > > ??? freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA > 2909/userBBB > > The call can be established and the data.wav file was generated without any > problem. However, once userAAA hung up, a segmentation fault occurred and > freeswitch was automatically shut down. Here are what I got from the > console: > > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA > 2909/userBBB > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > Ring-Ready skypiax/skypiax2/userAAA > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() > Channel [skypiax/skypiax2/userAAA] has been answered > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b > > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] > mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH->2909/userBBB > in context default > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > Ring-Ready skypiax/ANY/userBBB! > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() > Channel [skypiax/ANY/userBBB] has been answered > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA > [CS_DESTROY] > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] > Segmentation fault (core dumped) > > Please kindly let me know whether there's anything wrong with the dialplan > or the way how I originated the call. > > Thanks! > -Jingwei > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From asannucci at gmail.com Mon Jun 15 06:32:56 2009 From: asannucci at gmail.com (bakko) Date: Mon, 15 Jun 2009 15:32:56 +0200 Subject: [Freeswitch-users] Asterisks to Freeswitch CALL REJECTED In-Reply-To: <24031735.post@talk.nabble.com> References: <24031735.post@talk.nabble.com> Message-ID: <5F950C77D31F44AC96320EDE62F052DE@voztovoice> Hi, if you understand spanish look at: http://www.freeswitch.es/node/61 Regards From anthony.minessale at gmail.com Mon Jun 15 06:59:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Jun 2009 08:59:13 -0500 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <4A361708.8020808@xpirio.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> Message-ID: <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> Did you actually use ESL with the php wrapper when you tried? You can do all those things from outbound event socket fairly easily. That mod_php you saw, never worked it was just a stub and it didn't actually ever work when the guy who added it totally disappeared, I removed it from tree. And you can still do event socket over localhost on the same box if you so choose. If you really want a mod_php it's entirely possible but it would probably cost you upwards of 5k in development costs. 2009/6/15 Christian L?schenkohl > hi > > thank you very much for your input > i can say for me that i realy tried hard to use the event socket library, > but untill now i can't use it like i used all the agi scripts or even > mod_perl now. > > what i do most - in examples, if the server get's an incomming call > > - find the right user for the number (not that easy because of did in > austria), > from database or file > - build the right dial string for the bridge application (here i miss all > the php > string functions most) > - unsing mod_php functions like setVariable, getVariable, answer, transfer, > sleep > (i don't see how to do this with the php esl) > - or i check if the number is part of a conferencing product and build the > right > conference setup > > i think this would also be possible with lua and luasql, but i developed > years with > phpagi und i'm very used to php in every kind of scripting or > how-to-get-a-solution > situation (since over 10 years now). > > for me in our setup it's also the highest goal to get the servers mostly > independent > of each other. i think nobody of our costumers should be unreachable > because a central > scripting/event server or also database server has gone away (as developers > this happens > more often as we would like it to :-)) > > do not get me wrong, freeswitch is very powerfull and in the near future it > will replace > nearly all of our asterisk servers. > > in combination with php the freeswitch plattform would be heaven for me > > i also thought Brian Fertig has some source written (as posted on > http://wiki.freeswitch.org/wiki/Mod_php), > in combination of the mod_python rewrite (page was last modified in june > 2007). > > br > > > On 2009-06-14 01:15, Nik Middleton wrote: > > I couldn?t agree more. We?re working with a group that are developing a > > massive PHP based music application. They are experts in PHP and MySQL > > but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP > > to communicate with the FS event socket, allows them to work on the > > areas they know best and not worry about the telephony side too much. > > We went the lua route, and don?t use the dial plan at all. My view is > > to keep all db access and processing out of FS as much as possible. With > > the event socket you simply don?t need to embed anything apart from the > > essentials. > > > > We are now processing 100,000+ call setups a day (4 hours) per server > > all using php scripts to drive the application. We may well ultimately > > use C++ instead of PHP for the event socket comms, but right now PHP > > does just fine. > > > > Regards > > > > ------------------------------------------------------------------------ > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > > *Michael Collins > > *Sent:* 13 June 2009 21:57 > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > Perhaps you should look at controlling calls via the FreeSWITCH event > > socket instead of from the dialplan. The nice thing about the event > > socket is that your call control can happen on a separate machine. There > > is a PHP module for the ESL (event socket library) and it would be > > relatively easy for you to get going. Here are some links to get you > > started: > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > If you absolutely MUST have call control with scripts inside of the > > dialplan then there simply is no better choice than Lua. You can learn > > Lua in a few hours, but getting mod_php finished and debugged will take > > time, money, and other resources that no one seems willing to spend. > > Here is some information to consider: > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > ) if you want to discuss this further. > > > > -MC (IRC: mercutioviz) > > > > 2009/6/13 Christian L?schenkohl > > > > > > hello > > > > i am working for an austrian voip carrier. > > for a few months i work with freeswitch and it is simply great. > > it solves our needs in many places (high volume, flexible, stable). > > the only thing i really miss is the avalibilty of php as a call control > > language. > > mod_php or mod_freehp are not compiling anymore and my c++ knowledge > isn't > > that good (or even there :-) ). > > i know there is perl, i also implemented some applications (conference > > system with provisioning, > > inbound call routing to our application servers, some tests as pbx), but > > what should i say - > > perl and me aren't compatible in the end. > > > > the greatest thing for us would be that mod_php comes alive again with > > the functional state > > of mod_perl > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > there is also a bounty entry about mod_php, to pay for this would also > > be an option and > > could be discussed. > > > > keep on with the excellent work and greetings from austria > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung& Entwicklung VoIP > > > > xpirio > > Telekommunikation& Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/a926c87a/attachment.html From brian at freeswitch.org Mon Jun 15 07:21:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 09:21:02 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> Message-ID: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> To: <"user" Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: > Hi, > > I'm on version 13524, call from zoiper is ok, but when call zoiper, > it keep rejecting calls, anyone can help? I'm seems always not the > right time join in IRC :( > > http://pastebin.freeswitch.org/9383 > > > Thanks. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/d8b5d39b/attachment-0001.html From dujinfang at gmail.com Mon Jun 15 07:37:23 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 22:37:23 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: Yes, I can reproduce this, FYI, I have another box runs 13272 with the same zoiper without any problem. You can login to our box, I will find you on IRC. On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e04a36ae/attachment.html From mitul at enterux.com Mon Jun 15 07:38:48 2009 From: mitul at enterux.com (Mitul Limbani) Date: Mon, 15 Jun 2009 20:08:48 +0530 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> Message-ID: Anthm, I actually compiled ESL on PHP but wasnt able to figure out how to use it, too little documentation. Can any one throw more light on ESL? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 15-Jun-09, at 19:29, Anthony Minessale wrote: > Did you actually use ESL with the php wrapper when you tried? > You can do all those things from outbound event socket fairly easily. > > That mod_php you saw, never worked it was just a stub and it didn't > actually ever work > when the guy who added it totally disappeared, I removed it from tree. > > And you can still do event socket over localhost on the same box if > you so choose. > > If you really want a mod_php it's entirely possible but it would > probably cost you upwards > of 5k in development costs. > > > 2009/6/15 Christian L?schenkohl > hi > > thank you very much for your input > i can say for me that i realy tried hard to use the event socket > library, > but untill now i can't use it like i used all the agi scripts or > even mod_perl now. > > what i do most - in examples, if the server get's an incomming call > > - find the right user for the number (not that easy because of did > in austria), > from database or file > - build the right dial string for the bridge application (here i > miss all the php > string functions most) > - unsing mod_php functions like setVariable, getVariable, answer, > transfer, sleep > (i don't see how to do this with the php esl) > - or i check if the number is part of a conferencing product and > build the right > conference setup > > i think this would also be possible with lua and luasql, but i > developed years with > phpagi und i'm very used to php in every kind of scripting or how-to- > get-a-solution > situation (since over 10 years now). > > for me in our setup it's also the highest goal to get the servers > mostly independent > of each other. i think nobody of our costumers should be unreachable > because a central > scripting/event server or also database server has gone away (as > developers this happens > more often as we would like it to :-)) > > do not get me wrong, freeswitch is very powerfull and in the near > future it will replace > nearly all of our asterisk servers. > > in combination with php the freeswitch plattform would be heaven for > me > > i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php > ), > in combination of the mod_python rewrite (page was last modified in > june 2007). > > br > > > On 2009-06-14 01:15, Nik Middleton wrote: > > I couldn?t agree more. We?re working with a group that are > developing a > > massive PHP based music application. They are experts in PHP and > MySQL > > but not in VOIP/Telephony. By tuning an abstraction layer that > uses PHP > > to communicate with the FS event socket, allows them to work on the > > areas they know best and not worry about the telephony side too > much. > > We went the lua route, and don?t use the dial plan at all. My vie > w is > > to keep all db access and processing out of FS as much as > possible. With > > the event socket you simply don?t need to embed anything apart fro > m the > > essentials. > > > > We are now processing 100,000+ call setups a day (4 hours) per > server > > all using php scripts to drive the application. We may well > ultimately > > use C++ instead of PHP for the event socket comms, but right now PHP > > does just fine. > > > > Regards > > > > > --- > --------------------------------------------------------------------- > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > > *Michael Collins > > *Sent:* 13 June 2009 21:57 > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > Perhaps you should look at controlling calls via the FreeSWITCH > event > > socket instead of from the dialplan. The nice thing about the event > > socket is that your call control can happen on a separate machine. > There > > is a PHP module for the ESL (event socket library) and it would be > > relatively easy for you to get going. Here are some links to get you > > started: > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > If you absolutely MUST have call control with scripts inside of the > > dialplan then there simply is no better choice than Lua. You can > learn > > Lua in a few hours, but getting mod_php finished and debugged will > take > > time, money, and other resources that no one seems willing to spend. > > Here is some information to consider: > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > ) if you want to discuss this further. > > > > -MC (IRC: mercutioviz) > > > > 2009/6/13 Christian L?schenkohl > > > > > > hello > > > > i am working for an austrian voip carrier. > > for a few months i work with freeswitch and it is simply great. > > it solves our needs in many places (high volume, flexible, stable). > > the only thing i really miss is the avalibilty of php as a call > control > > language. > > mod_php or mod_freehp are not compiling anymore and my c++ > knowledge isn't > > that good (or even there :-) ). > > i know there is perl, i also implemented some applications > (conference > > system with provisioning, > > inbound call routing to our application servers, some tests as > pbx), but > > what should i say - > > perl and me aren't compatible in the end. > > > > the greatest thing for us would be that mod_php comes alive again > with > > the functional state > > of mod_perl > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > there is also a bounty entry about mod_php, to pay for this would > also > > be an option and > > could be discussed. > > > > keep on with the excellent work and greetings from austria > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung& Entwicklung VoIP > > > > xpirio > > Telekommunikation& Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > --- > --------------------------------------------------------------------- > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/b0c2f4a7/attachment-0001.html From christian.loeschenkohl at xpirio.com Mon Jun 15 07:55:39 2009 From: christian.loeschenkohl at xpirio.com (=?windows-1252?Q?Christian_L=F6schenkohl?=) Date: Mon, 15 Jun 2009 16:55:39 +0200 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> Message-ID: <4A3660EB.4070203@xpirio.com> i tried to i think i tried everthing and looked closely to everything in libs/esl/php (of course i build it and included the ESL.php file) but i do not get the idea in complete, does i work in a client-server way or in inbound mode like i want to (that is exactly my point) no examples are there (i would put them in the wiki if i had one) some simple code i would expect wot work, but i doesn't execute("setVariable", "codec_string=PCMA"); $esl->execute("answer"); $esl->execute("sleep", "2"); $esl->execute("streamFile", "/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello"); $esl->execute("hangup", "16"); ?> can you please help me, what do i get wrong? br On 2009-06-15 15:59, Anthony Minessale wrote: > Did you actually use ESL with the php wrapper when you tried? > You can do all those things from outbound event socket fairly easily. > > That mod_php you saw, never worked it was just a stub and it didn't > actually ever work > when the guy who added it totally disappeared, I removed it from tree. > > And you can still do event socket over localhost on the same box if you > so choose. > > If you really want a mod_php it's entirely possible but it would > probably cost you upwards > of 5k in development costs. > > > 2009/6/15 Christian L?schenkohl > > > hi > > thank you very much for your input > i can say for me that i realy tried hard to use the event socket > library, > but untill now i can't use it like i used all the agi scripts or > even mod_perl now. > > what i do most - in examples, if the server get's an incomming call > > - find the right user for the number (not that easy because of did > in austria), > from database or file > - build the right dial string for the bridge application (here i > miss all the php > string functions most) > - unsing mod_php functions like setVariable, getVariable, answer, > transfer, sleep > (i don't see how to do this with the php esl) > - or i check if the number is part of a conferencing product and > build the right > conference setup > > i think this would also be possible with lua and luasql, but i > developed years with > phpagi und i'm very used to php in every kind of scripting or > how-to-get-a-solution > situation (since over 10 years now). > > for me in our setup it's also the highest goal to get the servers > mostly independent > of each other. i think nobody of our costumers should be unreachable > because a central > scripting/event server or also database server has gone away (as > developers this happens > more often as we would like it to :-)) > > do not get me wrong, freeswitch is very powerfull and in the near > future it will replace > nearly all of our asterisk servers. > > in combination with php the freeswitch plattform would be heaven for me > > i also thought Brian Fertig has some source written (as posted on > http://wiki.freeswitch.org/wiki/Mod_php), > in combination of the mod_python rewrite (page was last modified in > june 2007). > > br > > > On 2009-06-14 01:15, Nik Middleton wrote: > > I couldn?t agree more. We?re working with a group that are > developing a > > massive PHP based music application. They are experts in PHP and > MySQL > > but not in VOIP/Telephony. By tuning an abstraction layer that > uses PHP > > to communicate with the FS event socket, allows them to work on the > > areas they know best and not worry about the telephony side too much. > > We went the lua route, and don?t use the dial plan at all. My > view is > > to keep all db access and processing out of FS as much as > possible. With > > the event socket you simply don?t need to embed anything apart > from the > > essentials. > > > > We are now processing 100,000+ call setups a day (4 hours) per server > > all using php scripts to drive the application. We may well > ultimately > > use C++ instead of PHP for the event socket comms, but right now PHP > > does just fine. > > > > Regards > > > > > ------------------------------------------------------------------------ > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of > > *Michael Collins > > *Sent:* 13 June 2009 21:57 > > *To:* freeswitch-users at lists.freeswitch.org > > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > Perhaps you should look at controlling calls via the FreeSWITCH event > > socket instead of from the dialplan. The nice thing about the event > > socket is that your call control can happen on a separate > machine. There > > is a PHP module for the ESL (event socket library) and it would be > > relatively easy for you to get going. Here are some links to get you > > started: > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > If you absolutely MUST have call control with scripts inside of the > > dialplan then there simply is no better choice than Lua. You can > learn > > Lua in a few hours, but getting mod_php finished and debugged > will take > > time, money, and other resources that no one seems willing to spend. > > Here is some information to consider: > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > > ) if you want to discuss this further. > > > > -MC (IRC: mercutioviz) > > > > 2009/6/13 Christian L?schenkohl > > > >> > > > > hello > > > > i am working for an austrian voip carrier. > > for a few months i work with freeswitch and it is simply great. > > it solves our needs in many places (high volume, flexible, stable). > > the only thing i really miss is the avalibilty of php as a call > control > > language. > > mod_php or mod_freehp are not compiling anymore and my c++ > knowledge isn't > > that good (or even there :-) ). > > i know there is perl, i also implemented some applications > (conference > > system with provisioning, > > inbound call routing to our application servers, some tests as > pbx), but > > what should i say - > > perl and me aren't compatible in the end. > > > > the greatest thing for us would be that mod_php comes alive again > with > > the functional state > > of mod_perl > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > there is also a bounty entry about mod_php, to pay for this would > also > > be an option and > > could be discussed. > > > > keep on with the excellent work and greetings from austria > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung& Entwicklung VoIP > > > > xpirio > > Telekommunikation& Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From dujinfang at gmail.com Mon Jun 15 08:07:50 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 23:07:50 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: Brain, You are not on irc right now and it is midnight so I'm gona to sleep. however when I try to reproduce that I found it event didn't get to Zoiper. I use the same Zoiper login to two boxes at the same time, version 13272 is ok while the other isn't. I noticed there is an extra line on the log of version 13524: version 13272: 2009-06-15 22:54:14 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/ internal/839 SOFIA INIT 2009-06-15 22:54:14 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/ internal/839) State Change CS_INIT -> CS_ROUTING version 13524: 2009-06-15 22:50:46 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/ internal/637 SOFIA INIT 2009-06-15 22:50:46 [DEBUG] sofia_glue.c:1599 sofia_glue_do_invite() sip:637 at 192.168.1.27:5070 %3 Setting proxy route to sofia/internal/637 2009-06-15 22:50:46 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/ internal/637) State Change CS_INIT -> CS_ROUTING And here is a more detailed paste: http://pastebin.freeswitch.org/9386 Thank you taking time for this. If you need more detail I'd like to collect and can open ssh for further debug. 7. On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/4ae227da/attachment.html From anthony.minessale at gmail.com Mon Jun 15 08:07:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Jun 2009 10:07:36 -0500 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <4A3660EB.4070203@xpirio.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> Message-ID: <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> look at the perl examples they should translate to php as all the objects and methods are the same. Does anyone who uses ESL + scripting have any time to toss up some wiki pages? 2009/6/15 Christian L?schenkohl > i tried to > i think i tried everthing and looked closely to everything in libs/esl/php > (of > course i build it and included the ESL.php file) > > but i do not get the idea in complete, does i work in a client-server way > or > in inbound mode like i want to (that is exactly my point) > > no examples are there (i would put them in the wiki if i had one) > some simple code i would expect wot work, but i doesn't > > > require_once("ESL.php"); > $esl = new eslConnection('127.0.0.1', '8021', 'asgag243tsa'); > > $esl->execute("setVariable", "codec_string=PCMA"); > $esl->execute("answer"); > $esl->execute("sleep", "2"); > $esl->execute("streamFile", > "/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello"); > $esl->execute("hangup", "16"); > > ?> > > can you please help me, what do i get wrong? > > br > > > On 2009-06-15 15:59, Anthony Minessale wrote: > > Did you actually use ESL with the php wrapper when you tried? > > You can do all those things from outbound event socket fairly easily. > > > > That mod_php you saw, never worked it was just a stub and it didn't > > actually ever work > > when the guy who added it totally disappeared, I removed it from tree. > > > > And you can still do event socket over localhost on the same box if you > > so choose. > > > > If you really want a mod_php it's entirely possible but it would > > probably cost you upwards > > of 5k in development costs. > > > > > > 2009/6/15 Christian L?schenkohl > > > > > > hi > > > > thank you very much for your input > > i can say for me that i realy tried hard to use the event socket > > library, > > but untill now i can't use it like i used all the agi scripts or > > even mod_perl now. > > > > what i do most - in examples, if the server get's an incomming call > > > > - find the right user for the number (not that easy because of did > > in austria), > > from database or file > > - build the right dial string for the bridge application (here i > > miss all the php > > string functions most) > > - unsing mod_php functions like setVariable, getVariable, answer, > > transfer, sleep > > (i don't see how to do this with the php esl) > > - or i check if the number is part of a conferencing product and > > build the right > > conference setup > > > > i think this would also be possible with lua and luasql, but i > > developed years with > > phpagi und i'm very used to php in every kind of scripting or > > how-to-get-a-solution > > situation (since over 10 years now). > > > > for me in our setup it's also the highest goal to get the servers > > mostly independent > > of each other. i think nobody of our costumers should be unreachable > > because a central > > scripting/event server or also database server has gone away (as > > developers this happens > > more often as we would like it to :-)) > > > > do not get me wrong, freeswitch is very powerfull and in the near > > future it will replace > > nearly all of our asterisk servers. > > > > in combination with php the freeswitch plattform would be heaven for > me > > > > i also thought Brian Fertig has some source written (as posted on > > http://wiki.freeswitch.org/wiki/Mod_php), > > in combination of the mod_python rewrite (page was last modified in > > june 2007). > > > > br > > > > > > On 2009-06-14 01:15, Nik Middleton wrote: > > > I couldn?t agree more. We?re working with a group that are > > developing a > > > massive PHP based music application. They are experts in PHP and > > MySQL > > > but not in VOIP/Telephony. By tuning an abstraction layer that > > uses PHP > > > to communicate with the FS event socket, allows them to work on > the > > > areas they know best and not worry about the telephony side too > much. > > > We went the lua route, and don?t use the dial plan at all. My > > view is > > > to keep all db access and processing out of FS as much as > > possible. With > > > the event socket you simply don?t need to embed anything apart > > from the > > > essentials. > > > > > > We are now processing 100,000+ call setups a day (4 hours) per > server > > > all using php scripts to drive the application. We may well > > ultimately > > > use C++ instead of PHP for the event socket comms, but right now > PHP > > > does just fine. > > > > > > Regards > > > > > > > > > ------------------------------------------------------------------------ > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > > ] *On Behalf > Of > > > *Michael Collins > > > *Sent:* 13 June 2009 21:57 > > > *To:* freeswitch-users at lists.freeswitch.org > > > > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > > > Perhaps you should look at controlling calls via the FreeSWITCH > event > > > socket instead of from the dialplan. The nice thing about the > event > > > socket is that your call control can happen on a separate > > machine. There > > > is a PHP module for the ESL (event socket library) and it would be > > > relatively easy for you to get going. Here are some links to get > you > > > started: > > > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > > > If you absolutely MUST have call control with scripts inside of > the > > > dialplan then there simply is no better choice than Lua. You can > > learn > > > Lua in a few hours, but getting mod_php finished and debugged > > will take > > > time, money, and other resources that no one seems willing to > spend. > > > Here is some information to consider: > > > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > > > > ) if you want to discuss this further. > > > > > > -MC (IRC: mercutioviz) > > > > > > 2009/6/13 Christian L?schenkohl > > > > > > > >> > > > > > > hello > > > > > > i am working for an austrian voip carrier. > > > for a few months i work with freeswitch and it is simply great. > > > it solves our needs in many places (high volume, flexible, > stable). > > > the only thing i really miss is the avalibilty of php as a call > > control > > > language. > > > mod_php or mod_freehp are not compiling anymore and my c++ > > knowledge isn't > > > that good (or even there :-) ). > > > i know there is perl, i also implemented some applications > > (conference > > > system with provisioning, > > > inbound call routing to our application servers, some tests as > > pbx), but > > > what should i say - > > > perl and me aren't compatible in the end. > > > > > > the greatest thing for us would be that mod_php comes alive again > > with > > > the functional state > > > of mod_perl > > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > > there is also a bounty entry about mod_php, to pay for this would > > also > > > be an option and > > > could be discussed. > > > > > > keep on with the excellent work and greetings from austria > > > > > > -- > > > Ing. Christian L?schenkohl > > > Technische Leitung, Forschung& Entwicklung VoIP > > > > > > xpirio > > > Telekommunikation& Service GmbH > > > Lakeside B04 > > > 9020 Klagenfurt > > > Austria > > > > > > T +43 (0) 5 77 11 - 1000 > > > F +43 (0) 5 77 11 - 1002 > > > E christian.loeschenkohl at xpirio.com > > > > > > > > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/be55ea82/attachment-0001.html From william.suffill at gmail.com Mon Jun 15 08:12:22 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 15 Jun 2009 11:12:22 -0400 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> Message-ID: <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> Any suggestions of what would be a good example in PHP using ESL to document? I'll take a stab at writing something up this week but it would help to have some idea what would be useful. I've used it and got it working but rather document a generic real life example versus my unique use cases. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/5fea324b/attachment.html From anthony.minessale at gmail.com Mon Jun 15 08:27:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Jun 2009 10:27:33 -0500 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> Message-ID: <191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com> A good start would probably be: example of making an inbound connection from script to FS and execute a FSAPI command, like status or show channels. example of making an inbound connection and listening for events and printing them serialized. example of an outbound socket connection where the call is answered, a variable is set then perhaps play one of the pre-installed files and hangup. That last one could be demonstrated using a native socket server or by using ivrd, a little mini forking daemon I added to listen for socket outbound calls and determine a script from channel variables and call that script assuming to use stdin/stdout as the socket. (kinda like agi's) I think that if everyone pooled their experienced together you could probably produce a wrapper that would allow you to use some of your legacy agi code with ESL, naturally you would have to change the names of the apps and a few other things but there is a lot to build on here. I left this portiion of the system where it is so that the community and how it's most commonly used will drive the direction the top layer of code takes. On Mon, Jun 15, 2009 at 10:12 AM, William Suffill wrote: > Any suggestions of what would be a good example in PHP using ESL to > document? I'll take a stab at writing something up this week but it would > help to have some idea what would be useful. I've used it and got it working > but rather document a generic real life example versus my unique use cases. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1d0b2823/attachment.html From dujinfang at gmail.com Mon Jun 15 08:53:29 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 15 Jun 2009 23:53:29 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: <1322BB82-F5CC-47AD-B686-557CA2147EDC@gmail.com> Hi, the difference is the Contact where the "%3B" should be ";" . Is it configurable or a bug? 13272: Call-ID: OTg4NmRlNzY5OThmNzgwM2E3ZmRkYzVhNjVmODMyYjA. User: 839 at 192.168.1.15 Contact: "user" Agent: Zoiper rev.3065 Status: Registered(UDP)(unknown) EXP(2009-06-16 00:54:12) Host: pbx3.veecue.com IP: 192.168.1.27 Port: 5070 Auth-User: 839 Auth-Realm: 192.168.1.15 13524: Call-ID: NWJhYWY0YjJmMzdlNWQ4MWIwZjc2NGM5NjQzZDU3NTg. User: 637 at 192.168.1.16 Contact: "user" Agent: Zoiper rev.3065 Status: Registered(UDP-NAT)(unknown) EXP(2009-06-16 01:42:56) Host: pbx1.veecue.com IP: 192.168.1.27 Port: 5070 Auth-User: 637 Auth-Realm: 192.168.1.16 On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/94c56399/attachment.html From steve.kurzeja at gmail.com Mon Jun 15 03:17:26 2009 From: steve.kurzeja at gmail.com (Steve Kurzeja) Date: Mon, 15 Jun 2009 22:17:26 +1200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> Message-ID: <5f7152000906150317i451b9f7s5c7a8d4293f7c5d1@mail.gmail.com> On Sat, Jun 13, 2009 at 4:54 AM, Michael Giagnocavo wrote: > Well, Nextone for instance has a database the keeps most of the state of > calls, and it?s replicated between the two nodes. (I seem to recall the > database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, > the CDRs still get truncated when there?s any kind of switchover. > > > BTW in Nextone v4.0.x the GNU db is used for storing configuration data like storing routes & other bits which is then loaded into memory. Nextone 4.3 and above uses postgres for this configuration data. The actual call state information is stored in memory and replicated to the standby box via some custom network protocol. Stateful call migration would be a very useful feature in FS but I imagine its way down the roadmap. But as to the original question of live upgrades, having some form of load balancing proxy and then bleeding off traffic from the box you want to upgrade is the most feasible approach, as others have mentioned. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1cb0e5a7/attachment.html From saeedahmad1981 at gmail.com Mon Jun 15 09:31:28 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 15 Jun 2009 18:31:28 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <5f7152000906150317i451b9f7s5c7a8d4293f7c5d1@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org><45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com><6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> <5f7152000906150317i451b9f7s5c7a8d4293f7c5d1@mail.gmail.com> Message-ID: <41C915AED64E45869CD719DEEDB8D8BF@saeedlaptop> Yeah I was missing this word: SCM => Stateful Call Migration. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Kurzeja Sent: Monday, June 15, 2009 12:17 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques On Sat, Jun 13, 2009 at 4:54 AM, Michael Giagnocavo wrote: Well, Nextone for instance has a database the keeps most of the state of calls, and it's replicated between the two nodes. (I seem to recall the database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still get truncated when there's any kind of switchover. BTW in Nextone v4.0.x the GNU db is used for storing configuration data like storing routes & other bits which is then loaded into memory. Nextone 4.3 and above uses postgres for this configuration data. The actual call state information is stored in memory and replicated to the standby box via some custom network protocol. Stateful call migration would be a very useful feature in FS but I imagine its way down the roadmap. But as to the original question of live upgrades, having some form of load balancing proxy and then bleeding off traffic from the box you want to upgrade is the most feasible approach, as others have mentioned. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e9b050c1/attachment-0001.html From Richard.Lamkin at mettoni.com Mon Jun 15 11:27:52 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Mon, 15 Jun 2009 19:27:52 +0100 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to an inbound call ? Message-ID: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin Richard.lamkin at mettonigroup.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/5fa3920d/attachment.html From christian.loeschenkohl at xpirio.com Mon Jun 15 12:08:39 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 15 Jun 2009 21:08:39 +0200 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> Message-ID: <4A369C37.1040107@xpirio.com> hi could you provide me a simple example? - connect with esl - get uuid - set a variable (e.g. codec_string=PCMA) - answer the channel - playback a file the script ist called from ivrd, if i get it right in the dialplan it's with ivrd started as ./ivrd -h 127.0.0.1 -p 9999 -------------------- in my setup $esl->api("help") works and also $esl->sendRecv("api help") but $esl->execute() does nothing i use version 1.0.4pre8 if it is helpfull br On 2009-06-15 17:12, William Suffill wrote: > Any suggestions of what would be a good example in PHP using ESL to > document? I'll take a stab at writing something up this week but it > would help to have some idea what would be useful. I've used it and got > it working but rather document a generic real life example versus my > unique use cases. > > -- W > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From edpimentl at gmail.com Mon Jun 15 12:25:28 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 15 Jun 2009 15:25:28 -0400 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" Message-ID: <9dc4a1670906151225x39126633g4ba159d03dab47d0@mail.gmail.com> FYI Now you can directly connect to MySql via javascript http://jsext.webloji.net/ http://jsext.sourceforge.net/JSEXT1.Mysql.html http://www.brainonfire.net/blog/ssjs-on-ubuntu/ And if you wish you had a V8, check out Google SSJS Engine BTW: JS is now 7x faster than it last year. -E http://Gpro.ws http://twitter.com/edpimentl http://facebook.com/facevalu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/22a83214/attachment.html From ron.freeswitch at mcleodnet.com Mon Jun 15 12:50:14 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Mon, 15 Jun 2009 12:50:14 -0700 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> Message-ID: Something to consider is how long will be PSTN allow the call to remain un-answered. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Lamkin Sent: Monday, June 15, 2009 11:28 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin Richard.lamkin at mettonigroup.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/2524086e/attachment-0001.html From brian at freeswitch.org Mon Jun 15 13:24:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 15:24:02 -0500 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> Message-ID: Survey says ... "execute the ring_ready application" /b On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote: > Something to consider is how long will be PSTN allow the call to > remain un-answered. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Richard Lamkin > Sent: Monday, June 15, 2009 11:28 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to > aninbound call ? > > I have a setup where I have a variety of SIP inbound calls > (originated from PSTN) coming from a SIP provider. The SIP lines > are single lines registered with the provider. The provider is > running with a Nortel CS2K. > > I am putting together a simple event driven operator attendant > console and I would like to set up a call queuing system where the > incoming calls are not answered until an operator is ready to > accept a call. I want the operator to know that a call is in the > ringing Q and who it is from. I do not want to auto answer the call > and put them in a MOH Q because the originator will be charged as > soon as the call is answered. > > My question is how do I get a SIP 180 ringing to be sent to an > inbound call and put that call in a Q? The CS2k does convert > ringing on inbound calls to media towards the originator. I?ve > looked through the wiki for examples but not found what I need in > either in dial plan or fifo operations. > > Any help would be gratefully appreciated. > > Regards > > Richard Lamkin > Richard.lamkin at mettonigroup.com > > ************************************************************************* > Please consider the environment before printing this e-mail > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/cdf91f10/attachment.html From lon at kickasspixels.com Mon Jun 15 14:46:01 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 15 Jun 2009 14:46:01 -0700 Subject: [Freeswitch-users] Gateway conference handshake trouble Message-ID: <5d3e0dc60906151446h3258ca4cld28fe1ca9daef8bd@mail.gmail.com> Hi, We are trying to work freeswitch into an older system as a conference bridge. Our existing gateways can hand off the call and they pass some DTMF signals to route everything. Currently, the gateway sends * and then freeswitch returns a # to accept the call for the conference. When that transaction is done the gateway waits 2 seconds to send either a 0 or 1 and indicate if the caller is a moderator/admin of the conference. Everything works until we get to the moderator flag. That part never appears in the logs or debug info in the console. But the person on the call, can hit 0 or 1 and then enter the conference call correctly. If they don't it will timeout and drop them. >From what we know the gateway is sending the moderator DTMF flag, but freeswitch is only listening on the call for it. Any idea would appreciated. Lon From stevecrozz at gmail.com Mon Jun 15 14:51:30 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 15 Jun 2009 14:51:30 -0700 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <9dc4a1670906151225x39126633g4ba159d03dab47d0@mail.gmail.com> References: <9dc4a1670906151225x39126633g4ba159d03dab47d0@mail.gmail.com> Message-ID: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> What a great project, does anyone know what's needed to make these libraries available to freeswitch scripts? --Stephen On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl wrote: > FYI > Now you can directly connect to MySql via javascript > > http://jsext.webloji.net/ > http://jsext.sourceforge.net/JSEXT1.Mysql.html > http://www.brainonfire.net/blog/ssjs-on-ubuntu/ > > And if you wish you had a V8, check out Google SSJS Engine > BTW: JS is now 7x faster than it last year. > > -E > http://Gpro.ws > http://twitter.com/edpimentl > http://facebook.com/facevalu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/2e15f5cd/attachment.html From krice at freeswitch.org Mon Jun 15 14:58:46 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Jun 2009 16:58:46 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> Message-ID: You know you can already access any sql support by UnixODBC via mod_spidermonkey already... And NO its not slow (maybe that was true 10 years ago, but not any longer) From: Stephen Crosby Reply-To: Date: Mon, 15 Jun 2009 14:51:30 -0700 To: Subject: Re: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" What a great project, does anyone know what's needed to make these libraries available to freeswitch scripts? --Stephen On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl wrote: > FYI > Now you can directly connect to MySql via javascript > > http://jsext.webloji.net/ > http://jsext.sourceforge.net/JSEXT1.Mysql.html > http://www.brainonfire.net/blog/ssjs-on-ubuntu/ > > And if you wish you had a V8, check out Google SSJS Engine > BTW: JS is now 7x faster than it last year. > > -E > http://Gpro.ws > http://twitter.com/edpimentl > http://facebook.com/facevalu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/d7532d6b/attachment-0001.html From stevecrozz at gmail.com Mon Jun 15 15:08:12 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 15 Jun 2009 15:08:12 -0700 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> Message-ID: <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> I'm actually much more interested in the HTTP library and a few other components than MySQL. Freeswitch's spidermonkey CURL library doesn't provide returned HTTP status codes and JSEXT does. That said, I'm still somewhat interested in the mysql library over odbc. For me, the only thing I've ever really done with ODBC is use it to bridge the gap between freeswitch mod_spidermonkey and mysql server. I think it would be nice to not need it. --Stephen On Mon, Jun 15, 2009 at 2:58 PM, Ken Rice wrote: > You know you can already access any sql support by UnixODBC via > mod_spidermonkey already... And NO its not slow (maybe that was true 10 > years ago, but not any longer) > > > ------------------------------ > *From: *Stephen Crosby > *Reply-To: * > *Date: *Mon, 15 Jun 2009 14:51:30 -0700 > *To: * > *Subject: *Re: [Freeswitch-users] Access MySQL directly via Javascript > using SSJS Engines .... "really!" > > > What a great project, does anyone know what's needed to make these > libraries available to freeswitch scripts? > > --Stephen > > On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl wrote: > > FYI > Now you can directly connect to MySql via javascript > > http://jsext.webloji.net/ > http://jsext.sourceforge.net/JSEXT1.Mysql.html > http://www.brainonfire.net/blog/ssjs-on-ubuntu/ > > And if you wish you had a V8, check out Google SSJS Engine > BTW: JS is now 7x faster than it last year. > > -E > http://Gpro.ws > http://twitter.com/edpimentl > http://facebook.com/facevalu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e041de66/attachment.html From brian at freeswitch.org Mon Jun 15 15:23:04 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 17:23:04 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> Message-ID: <538DFEC7-F2BB-45C0-837D-A5DB0B9F889D@freeswitch.org> On Jun 15, 2009, at 5:08 PM, Stephen Crosby wrote: > I'm actually much more interested in the HTTP library and a few > other components than MySQL. Freeswitch's spidermonkey CURL library > doesn't provide returned HTTP status codes and JSEXT does.\\\ Patch it! ;) > > > That said, I'm still somewhat interested in the mysql library over > odbc. For me, the only thing I've ever really done with ODBC is use > it to bridge the gap between freeswitch mod_spidermonkey and mysql > server. I think it would be nice to not need it. > > --Stephen From edpimentl at gmail.com Mon Jun 15 15:45:47 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 15 Jun 2009 18:45:47 -0400 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> Message-ID: <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> Actually these new SSJS engines such GoogleV8 and other such as JAXER Bring a entire new way of building robust webapp/desktop app/ mobile app like it has never been built before... For those that love Google GWT=Java_To_Javascript and dislike verbosity of Java, there is PyJamas ... Google GWT=Python_To_Javascript JS today is not the same it was years ago or even months ago. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/ce481122/attachment.html From brian at freeswitch.org Mon Jun 15 16:35:00 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 18:35:00 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> Message-ID: <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> I don't think V8 will work on 64bit yet will it? /b On Jun 15, 2009, at 5:45 PM, EdPimentl wrote: > Actually these new SSJS engines such GoogleV8 and other such as JAXER > Bring a entire new way of building robust webapp/desktop app/ mobile > app like it has never been built before... > > For those that love Google GWT=Java_To_Javascript and dislike > verbosity of Java, > there is PyJamas ... Google GWT=Python_To_Javascript > > JS today is not the same it was years ago or even months ago. > > -E From Prometheus001 at gmx.net Mon Jun 15 17:19:57 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 02:19:57 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? Message-ID: <4A36E52D.2010005@gmx.net> I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter From brian at freeswitch.org Mon Jun 15 17:27:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 19:27:44 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A36E52D.2010005@gmx.net> References: <4A36E52D.2010005@gmx.net> Message-ID: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: > > What is the best way to have this done? Move the call to park and then > retransfer again with intercom, or is there a better solution? From darklion11 at yahoo.com Mon Jun 15 18:31:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 18:31:13 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <7E486E4D-B08F-4BB3-A1D2-FD97DBACF204@gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <7E486E4D-B08F-4BB3-A1D2-FD97DBACF204@gmail.com> Message-ID: <24045733.post@talk.nabble.com> I like to use this GUI for both OS windows and linux. Wikipbx and PFSENSE is for linux only... Just a simple website that I need to integrate... seven-8 wrote: > > > On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote: > >> >> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >> fully >> developed... >> Is there any GUI with billing options? >> >> > > AFAIK, no fully developed GUI available yet, just curious, why are you > finding a GUI instead of wikipbx or pfsense? > > >> seven-8 wrote: >>> >>> http://www.tcapi.org/index.php?title=Main_Page >>> >>> >>> >>> On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>> >>>> >>>> is there any available gui for freeswitch using cake php complete >>>> instead of >>>> wikipbx, spice softphone or pfsense? >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24045733.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 18:38:18 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 18:38:18 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24045824.post@talk.nabble.com> I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 18:43:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 18:43:41 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Message-ID: <24045890.post@talk.nabble.com> Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER = root PASS = password ..... I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24045890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Jun 15 19:28:23 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 15 Jun 2009 21:28:23 -0500 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <1245058387.4694.5.camel@dk-d820> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> <1245058387.4694.5.camel@dk-d820> Message-ID: <13DE8568-1F4D-4224-B07B-77C03B844726@freeswitch.org> Hehe, where can I buy stock in this company? :) -MC Sent from my iPhone On Jun 15, 2009, at 4:33 AM, David Knell wrote: > On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote: >> What is the current status of Freeswitch? Can I safely use it in a >> large scale commercial environment? How active is the Freeswitch >> developer community? > > Hi Paul - > > We've used FS over the last 18 months or so to handle millions of > calls > - some wholesale in/out, some IVR, some calling card, some > callthrough - > with a total value in the millions of dollars; we have no complaints. > > --Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From diego.viola at gmail.com Mon Jun 15 19:56:31 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 15 Jun 2009 22:56:31 -0400 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24032171.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> Message-ID: <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz wrote: > > Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully > developed... > Is there any GUI with billing options? > > > seven-8 wrote: > > > > http://www.tcapi.org/index.php?title=Main_Page > > > > > > > > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: > > > >> > >> is there any available gui for freeswitch using cake php complete > >> instead of > >> wikipbx, spice softphone or pfsense? > >> -- > >> View this message in context: > >> > http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/d243308c/attachment.html From edpimentl at gmail.com Mon Jun 15 19:58:09 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 15 Jun 2009 22:58:09 -0400 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> Message-ID: <9dc4a1670906151958o4914b94cr46ac4ba313ba7315@mail.gmail.com> As of April 09 it did not support 64bit .... not sure if it has been added since then. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1325a8ba/attachment.html From darklion11 at yahoo.com Mon Jun 15 20:06:00 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 20:06:00 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> Message-ID: <24046873.post@talk.nabble.com> Can you share me the link of it so i can try... Please Diego Viola wrote: > > I'm currently writing a rails app that uses mod_nibblebill for billing, > it's > a calling card app. > > Diego > > On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz wrote: > >> >> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully >> developed... >> Is there any GUI with billing options? >> >> >> seven-8 wrote: >> > >> > http://www.tcapi.org/index.php?title=Main_Page >> > >> > >> > >> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >> > >> >> >> >> is there any available gui for freeswitch using cake php complete >> >> instead of >> >> wikipbx, spice softphone or pfsense? >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 15 22:14:40 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 00:14:40 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <9dc4a1670906151958o4914b94cr46ac4ba313ba7315@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> <9dc4a1670906151958o4914b94cr46ac4ba313ba7315@mail.gmail.com> Message-ID: <30E63C81-EAAF-407B-9B50-E05BD5EB1B1C@freeswitch.org> Pretty useless without 64bit support. /b On Jun 15, 2009, at 9:58 PM, EdPimentl wrote: > As of April 09 it did not support 64bit .... not sure if it has been > added since then. > -E From darklion11 at yahoo.com Mon Jun 15 23:08:39 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 23:08:39 -0700 (PDT) Subject: [Freeswitch-users] created external5090 on profile not working? Message-ID: <24048269.post@talk.nabble.com> I created a profile name external5090 on /usr/local/freeswitch/conf/sip_profiles/external5090.xml... Change ext-sip-ip and ext-rtp-ip for a server 192.168.0.104 with sip-port: 5090... My local Ip is 192.168.0.105... I see it with I type it on the API freeswitch and type sofia status is there... How can I know that it is working? can u send me a API freeswitch for it? may code is originate sofia/external5090/1002 at 192.168.0.104:5090 5090 is this correct? -- View this message in context: http://www.nabble.com/created-external5090-on-profile-not-working--tp24048269p24048269.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From durk.debeer at isp.solcon.nl Mon Jun 15 23:38:22 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Tue, 16 Jun 2009 08:38:22 +0200 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: References: Message-ID: >> >> Hello I've minimized de xml files where possible to make a dialplan that is >> as short as possible. Now do I've this funny effect to dial my extensions >> who are running from 200 to 207. It seams that I'm able to dial an >> extension in closed in a number. So for instants if I dial 120275 extension >> 202 will ring even tried it whit two extensions in a number like 202205 . >> This results in the first extension ringing so 202205, 202 will ring >> 205202, 205 will ring. At this time I'm unable to pinpoint the cause of >> this behaviour. Could someone point me to the cause of this effect > I don't understand the problem, but my general advice is this: learn to read > the FreeSWITCH logs carefully. Make sure that the log level is set to "debug", > as it is in the default configuration, then carefully check the log files to > see which dialplan extension matched and how the call was processed. After reading this, a colleague of mine had a look at the logs and found out that we had goofed up the regular expressions in the dialplan. This made Freeswitch dial the number of an extension if its sequence was found in the dialed number so lets say the extension has number 202 and the number dialed was 15320264 it would find the 202 sequence in the dialed number and then it would dial the 202 extension. So it seems this one is a stupid mistake of us. Thanks to all that responded \d From darklion11 at yahoo.com Tue Jun 16 00:44:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 00:44:31 -0700 (PDT) Subject: [Freeswitch-users] Is there anyone who is connected to PCCW? Message-ID: <24049302.post@talk.nabble.com> PCCW is use for making calls through IP connected through cellphone just enter the areacode for example 900639274522123 900-prefix 63-areacode 9274522123 - number? Has anyone has tried it? Please help me how to connect to it -- View this message in context: http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Tue Jun 16 01:15:51 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 16 Jun 2009 16:15:51 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> Message-ID: <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> Hi Giovanni, I've reported it in Jira. Here's the bug url: http://jira.freeswitch.org/browse/MODSKYPIAX-35 Thanks, -Jingwei On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli wrote: > Hi Jingwel, > thanks for reporting. > > Could you please add a Jira issue with as much details as possible? > > general guide for reporting bugs: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > what to add for skypiax: > > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests > > mod_skypiax Jira: > http://jira.freeswitch.org/browse/MODSKYPIAX > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang > wrote: > > Hi Team, > > > > I've been using the record_session feature to record call sessions. > Here's > > how I prepared the dialplan: > > > > > > > > > > > > > > > > > > And here's how I trigger it: > > > > freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA > > 2909/userBBB > > > > The call can be established and the data.wav file was generated without > any > > problem. However, once userAAA hung up, a segmentation fault occurred and > > freeswitch was automatically shut down. Here are what I got from the > > console: > > > > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA > > 2909/userBBB > > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 > switch_channel_set_name() > > New Channel skypiax/skypiax2/userAAA > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] > > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > > Ring-Ready skypiax/skypiax2/userAAA > > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 > outbound_channel_answered() > > Channel [skypiax/skypiax2/userAAA] has been answered > > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 > switch_ivr_session_transfer() > > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] > > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: > > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b > > > > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] > > mod_dialplan_xml.c:252 dialplan_hunt() Processing > FreeSWITCH->2909/userBBB > > in context default > > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 > switch_channel_set_name() > > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] > > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > > Ring-Ready skypiax/ANY/userBBB! > > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 > outbound_channel_answered() > > Channel [skypiax/ANY/userBBB] has been answered > > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 > > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA > [CS_EXECUTE] > > [NORMAL_CLEARING] > > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 > > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA > > [CS_DESTROY] > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > > switch_core_session_thread() Close Channel skypiax/ANY/userBBB > [CS_DESTROY] > > Segmentation fault (core dumped) > > > > Please kindly let me know whether there's anything wrong with the > dialplan > > or the way how I originated the call. > > > > Thanks! > > -Jingwei > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/bb8fe2e9/attachment-0001.html From gmaruzz at celliax.org Tue Jun 16 01:42:53 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 16 Jun 2009 10:42:53 +0200 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> Message-ID: <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> Hi Jingwei, Thanks a lot! I'll take care of as soon as possible. Btw, before I read the Jira, are you testing in linux? If you are testing on linux, would you please report how it is performing under load? I mean, what is the load average with, let say, 10 or 20 or more concurrent Skype call? This has nothing to do with your bug, but will help me in getting better performances. Ciao for now, and thanks again for reporting! -giovanni On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang wrote: > Hi Giovanni, > > I've reported it in Jira. Here's the bug url: > > http://jira.freeswitch.org/browse/MODSKYPIAX-35 > > Thanks, > -Jingwei > > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli > wrote: >> >> Hi Jingwel, >> thanks for reporting. >> >> Could you please add a Jira issue with as much details as possible? >> >> general guide for reporting bugs: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> what to add for skypiax: >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >> >> mod_skypiax Jira: >> http://jira.freeswitch.org/browse/MODSKYPIAX >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang >> wrote: >> > Hi Team, >> > >> > I've been using the record_session feature to record call sessions. >> > Here's >> > how I prepared the dialplan: >> > >> > ??? >> > ????? >> > ??????? >> > ??????? >> > ????? >> > ??? >> > >> > And here's how I trigger it: >> > >> > ??? freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA >> > 2909/userBBB >> > >> > The call can be established and the data.wav file was generated without >> > any >> > problem. However, once userAAA hung up, a segmentation fault occurred >> > and >> > freeswitch was automatically shut down. Here are what I got from the >> > console: >> > >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA >> > 2909/userBBB >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >> > switch_channel_set_name() >> > New Channel skypiax/skypiax2/userAAA >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >> > remote_party_is_ringing() >> > Ring-Ready skypiax/skypiax2/userAAA >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >> > outbound_channel_answered() >> > Channel [skypiax/skypiax2/userAAA] has been answered >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >> > switch_ivr_session_transfer() >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >> > >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >> > FreeSWITCH->2909/userBBB >> > in context default >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >> > switch_channel_set_name() >> > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >> > remote_party_is_ringing() >> > Ring-Ready skypiax/ANY/userBBB! >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >> > outbound_channel_answered() >> > Channel [skypiax/ANY/userBBB] has been answered >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >> > [CS_EXECUTE] >> > [NORMAL_CLEARING] >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >> > [CS_DESTROY] >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >> > [CS_DESTROY] >> > Segmentation fault (core dumped) >> > >> > Please kindly let me know whether there's anything wrong with the >> > dialplan >> > or the way how I originated the call. >> > >> > Thanks! >> > -Jingwei >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Tue Jun 16 02:17:00 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 05:17:00 -0400 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24046873.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> Message-ID: <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: > > Can you share me the link of it so i can try... Please > > Diego Viola wrote: >> >> I'm currently writing a rails app that uses mod_nibblebill for billing, >> it's >> a calling card app. >> >> Diego >> >> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz wrote: >> >>> >>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully >>> developed... >>> Is there any GUI with billing options? >>> >>> >>> seven-8 wrote: >>> > >>> > http://www.tcapi.org/index.php?title=Main_Page >>> > >>> > >>> > >>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>> > >>> >> >>> >> is there any available gui for freeswitch using cake php complete >>> >> instead of >>> >> wikipbx, spice softphone or pfsense? >>> >> -- >>> >> View this message in context: >>> >> >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >> >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Tue Jun 16 02:24:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 11:24:11 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: References: <4A36E52D.2010005@gmx.net> Message-ID: <4A3764BB.3040403@gmx.net> Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: > click on the AA button? :) > > /b > > On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: > > >> What is the best way to have this done? Move the call to park and then >> retransfer again with intercom, or is there a better solution? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Tue Jun 16 02:26:29 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 02:26:29 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> Message-ID: <24050713.post@talk.nabble.com> Thanks for that info... Can you send me this project if and only if it is already finished on this email darklion at yahoo.com? Thanks a lot... Diego Viola wrote: > > I'm currently rewriting the entire thing, it was a commercial app > first, but I'm re-writing it in order to make it open source. It's not > ready yet, as soon as I finish it, I will release it to the public. > > Diego > > On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: >> >> Can you share me the link of it so i can try... Please >> >> Diego Viola wrote: >>> >>> I'm currently writing a rails app that uses mod_nibblebill for billing, >>> it's >>> a calling card app. >>> >>> Diego >>> >>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>> wrote: >>> >>>> >>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>> fully >>>> developed... >>>> Is there any GUI with billing options? >>>> >>>> >>>> seven-8 wrote: >>>> > >>>> > http://www.tcapi.org/index.php?title=Main_Page >>>> > >>>> > >>>> > >>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>> > >>>> >> >>>> >> is there any available gui for freeswitch using cake php complete >>>> >> instead of >>>> >> wikipbx, spice softphone or pfsense? >>>> >> -- >>>> >> View this message in context: >>>> >> >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Tue Jun 16 02:32:21 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 16 Jun 2009 17:32:21 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> Message-ID: <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> Sure, I'll append to you the result tomorrow. Regards, -Jingwei On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli wrote: > Hi Jingwei, > > Thanks a lot! I'll take care of as soon as possible. > > Btw, before I read the Jira, are you testing in linux? > > If you are testing on linux, would you please report how it is > performing under load? I mean, what is the load average with, let say, > 10 or 20 or more concurrent Skype call? > > This has nothing to do with your bug, but will help me in getting > better performances. > > Ciao for now, and thanks again for reporting! > > -giovanni > > > > > On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang > wrote: > > Hi Giovanni, > > > > I've reported it in Jira. Here's the bug url: > > > > http://jira.freeswitch.org/browse/MODSKYPIAX-35 > > > > Thanks, > > -Jingwei > > > > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Hi Jingwel, > >> thanks for reporting. > >> > >> Could you please add a Jira issue with as much details as possible? > >> > >> general guide for reporting bugs: > >> http://wiki.freeswitch.org/wiki/Reporting_Bugs > >> > >> what to add for skypiax: > >> > >> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests > >> > >> mod_skypiax Jira: > >> http://jira.freeswitch.org/browse/MODSKYPIAX > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> > >> > >> > >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang > >> wrote: > >> > Hi Team, > >> > > >> > I've been using the record_session feature to record call sessions. > >> > Here's > >> > how I prepared the dialplan: > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > And here's how I trigger it: > >> > > >> > freeswitch at localhost.localdomain>originate > skypiax/skypiax2/userAAA > >> > 2909/userBBB > >> > > >> > The call can be established and the data.wav file was generated > without > >> > any > >> > problem. However, once userAAA hung up, a segmentation fault occurred > >> > and > >> > freeswitch was automatically shut down. Here are what I got from the > >> > console: > >> > > >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA > >> > 2909/userBBB > >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 > >> > switch_channel_set_name() > >> > New Channel skypiax/skypiax2/userAAA > >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] > >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 > >> > remote_party_is_ringing() > >> > Ring-Ready skypiax/skypiax2/userAAA > >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 > >> > outbound_channel_answered() > >> > Channel [skypiax/skypiax2/userAAA] has been answered > >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 > >> > switch_ivr_session_transfer() > >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] > >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: > >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b > >> > > >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] > >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing > >> > FreeSWITCH->2909/userBBB > >> > in context default > >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 > >> > switch_channel_set_name() > >> > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] > >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 > >> > remote_party_is_ringing() > >> > Ring-Ready skypiax/ANY/userBBB! > >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 > >> > outbound_channel_answered() > >> > Channel [skypiax/ANY/userBBB] has been answered > >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 > >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA > >> > [CS_EXECUTE] > >> > [NORMAL_CLEARING] > >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 > >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB > >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) > Ended > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA > >> > [CS_DESTROY] > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB > >> > [CS_DESTROY] > >> > Segmentation fault (core dumped) > >> > > >> > Please kindly let me know whether there's anything wrong with the > >> > dialplan > >> > or the way how I originated the call. > >> > > >> > Thanks! > >> > -Jingwei > >> > > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/d090ae23/attachment-0001.html From diego.viola at gmail.com Tue Jun 16 03:01:12 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 06:01:12 -0400 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24050713.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> <24050713.post@talk.nabble.com> Message-ID: <86a32abc0906160301p802a71arb8d67bc9a0ebcfa9@mail.gmail.com> Sure, I will let you know when it's done. On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruz wrote: > > Thanks for that info... Can you send me this project if and only if it is > already finished on this email darklion at yahoo.com? Thanks a lot... > > > Diego Viola wrote: >> >> I'm currently rewriting the entire thing, it was a commercial app >> first, but I'm re-writing it in order to make it open source. It's not >> ready yet, as soon as I finish it, I will release it to the public. >> >> Diego >> >> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: >>> >>> Can you share me the link of it so i can try... Please >>> >>> Diego Viola wrote: >>>> >>>> I'm currently writing a rails app that uses mod_nibblebill for billing, >>>> it's >>>> a calling card app. >>>> >>>> Diego >>>> >>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>>> fully >>>>> developed... >>>>> Is there any GUI with billing options? >>>>> >>>>> >>>>> seven-8 wrote: >>>>> > >>>>> > http://www.tcapi.org/index.php?title=Main_Page >>>>> > >>>>> > >>>>> > >>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>>> > >>>>> >> >>>>> >> is there any available gui for freeswitch using cake php complete >>>>> >> instead of >>>>> >> wikipbx, spice softphone or pfsense? >>>>> >> -- >>>>> >> View this message in context: >>>>> >> >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Tue Jun 16 03:20:06 2009 From: dujinfang at gmail.com (seven) Date: Tue, 16 Jun 2009 18:20:06 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: Hi brain, Are you still looking into this? I think it must be some error when it register, I manually changed the contract str in the registration db, immediately it works. After re- register, stop work again. Should I report this to jira? sqlite> select contact from sip_registrations where contact like '%637%'; contact "user" sqlite> update sip_registrations set contact='"user" ' where contact like '%637%'; On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/80a991b7/attachment.html From darklion11 at yahoo.com Tue Jun 16 03:59:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 03:59:01 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24050713.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> <24050713.post@talk.nabble.com> Message-ID: <24051970.post@talk.nabble.com> Hello sir, Do you know how to connect to two freeswitch at a time with different Ip addresses? If a user is register on FreeSwitch 1, the user should not have another account or he/she will not register anymore for Freeswitch 2? They can call each other... I already make one but an error occur Can't find user 1566331 at 192.168.0.105 You must define a domain called "192.168.0.105" in your directory and add a user="1566331" ..... Can you give me an example? Thanks for the help. Edmar Cruz wrote: > > Thanks for that info... Can you send me this project if and only if it is > already finished on this email darklion at yahoo.com? Thanks a lot... > > > Diego Viola wrote: >> >> I'm currently rewriting the entire thing, it was a commercial app >> first, but I'm re-writing it in order to make it open source. It's not >> ready yet, as soon as I finish it, I will release it to the public. >> >> Diego >> >> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: >>> >>> Can you share me the link of it so i can try... Please >>> >>> Diego Viola wrote: >>>> >>>> I'm currently writing a rails app that uses mod_nibblebill for billing, >>>> it's >>>> a calling card app. >>>> >>>> Diego >>>> >>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>>> fully >>>>> developed... >>>>> Is there any GUI with billing options? >>>>> >>>>> >>>>> seven-8 wrote: >>>>> > >>>>> > http://www.tcapi.org/index.php?title=Main_Page >>>>> > >>>>> > >>>>> > >>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>>> > >>>>> >> >>>>> >> is there any available gui for freeswitch using cake php complete >>>>> >> instead of >>>>> >> wikipbx, spice softphone or pfsense? >>>>> >> -- >>>>> >> View this message in context: >>>>> >> >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24051970.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue Jun 16 04:35:02 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 06:35:02 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> If you can catch brian or me on irc can you provide remote access to this box and we should be able to fix this pretty quick Mike On Jun 16, 2009, at 5:20 AM, seven wrote: > Hi brain, > > Are you still looking into this? > > I think it must be some error when it register, I manually changed > the contract str in the registration db, immediately it works. > After re-register, stop work again. > > Should I report this to jira? > > sqlite> select contact from sip_registrations where contact like > '%637%'; > contact > "user" 637@ > 192.168.1.27: > 5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip > %3A637%40192.168.1.27%3A5070%3Brinstance > %3D8df223525ea557b0%3Btransport%3DUDP> > > sqlite> update sip_registrations set contact='"user" :5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip > %3A637%40192.168.1.27%3A5070>' where contact like '%637%'; > > > > > On Jun 15, 2009, at 10:21 PM, Brian West wrote: >> To: <"user" >> >> Can you reproduce this or let us in your box to look at it... >> someone else reported this but I have yet to be able to reproduce it. >> >> /b >> >> On Jun 15, 2009, at 2:41 AM, seven wrote: >> >>> Hi, >>> >>> I'm on version 13524, call from zoiper is ok, but when call >>> zoiper, it keep rejecting calls, anyone can help? I'm seems always >>> not the right time join in IRC :( >>> >>> http://pastebin.freeswitch.org/9383 >>> >>> >>> Thanks. >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/50718e0b/attachment-0001.html From mike at jerris.com Tue Jun 16 04:50:39 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 06:50:39 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A3764BB.3040403@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> Message-ID: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX wrote: > Hello Brian, > > this is too easy :-). > > This is for a small callcenter app and I only want the user to pickup > the call once (to accept the call in X-Lite (or a Snom phone) and to > start the workflow on the web application). I do not want him to > accept > the call on the phone and then on the Web app. > > Best regards > Peter > > > > Brian West schrieb: >> click on the AA button? :) >> >> /b >> >> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >> >> >>> What is the best way to have this done? Move the call to park and >>> then >>> retransfer again with intercom, or is there a better solution? >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Tue Jun 16 04:55:02 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 16 Jun 2009 07:55:02 -0400 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A3764BB.3040403@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> Message-ID: <4A378816.8020504@freeswitch.org> Peter P GMX wrote: > Hello Brian, > > this is too easy :-). > > This is for a small callcenter app and I only want the user to pickup > the call once (to accept the call in X-Lite (or a Snom phone) and to > start the workflow on the web application). I do not want him to accept > the call on the phone and then on the Web app. > is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray From gcd at i.ph Tue Jun 16 05:02:58 2009 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 16 Jun 2009 20:02:58 +0800 Subject: [Freeswitch-users] Is there anyone who is connected to PCCW? In-Reply-To: <24049302.post@talk.nabble.com> References: <24049302.post@talk.nabble.com> Message-ID: <7d0bfd8c0906160502u7d3f6decxa1dc667a44eb6c4@mail.gmail.com> what is PCCW? could you please fill in more details what you like to do. to connect mobile phones w/ FS, the mobile phone has to have SIP feature. pls search the Wiki for some models. -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Tue, Jun 16, 2009 at 3:44 PM, Edmar Cruz wrote: > > PCCW is use for making calls through IP connected through cellphone just > enter the areacode for example > > 900639274522123 > > 900-prefix > 63-areacode > 9274522123 - number? > > Has anyone has tried it? > > Please help me how to connect to it > -- > View this message in context: > http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/9c8171e0/attachment.html From paul.degt at gmail.com Tue Jun 16 05:18:24 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Tue, 16 Jun 2009 08:18:24 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A35C72A.6030804@gmail.com> References: <4A35C72A.6030804@gmail.com> Message-ID: <4A378D90.2040209@gmail.com> Solved by replacing "auto-nat" with public ip in public profile "external_sip-ip" and "extrenal-rtp-ip" params. I believe values for these params used to be taken from vars.xml and so would have public ips by default - would be nice to document such changes in README. paul.degt at gmail.com wrote: > Trying out latest trunk ans seems like js function session.getDigits() > stopped working (not collecting any digits), I do see > > switch_rtp.c:1560 Send end packet for [5] ts=2222260 > dur=2080/2080/2000 seq=8732 > > in debug log so I assume dtmf is ok. > Anybody can shed some light on why wouldn't it work now? > Works just fine under 1.0.3 release. I use slightly modified version > of disa.js from fs examples. > > Thanks. > From Prometheus001 at gmx.net Tue Jun 16 05:32:29 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 14:32:29 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A378816.8020504@freeswitch.org> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A378816.8020504@freeswitch.org> Message-ID: <4A3790DD.8090009@gmx.net> Hello Ray, I do use event socket and it pushes me a link on the website whenever a call for this agent comes in. It's just a matter of visibility. The agent may still finish his old workflow and is still entering data. When a call comes in then and he picks up the phone, the data he just entered is gone away. So I would like the web app to drive answering the call. It gives a better visibility about what he is doing to the callcenter agent. Best regards Peter Raymond Chandler schrieb: > Peter P GMX wrote: > >> Hello Brian, >> >> this is too easy :-). >> >> This is for a small callcenter app and I only want the user to pickup >> the call once (to accept the call in X-Lite (or a Snom phone) and to >> start the workflow on the web application). I do not want him to accept >> the call on the phone and then on the Web app. >> >> > is there any reason you don't make your web app listen to event socket > or event sink to catch the answer event and start the workflow? then you > just need to answer the call on the softphone and the webapp should > automatically start the workflow. > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Tue Jun 16 05:38:21 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 14:38:21 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> Message-ID: <4A37923D.8060809@gmx.net> Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: > The only way I can think to do this today would be to cancel the call > and re send with the intercom headers for a phone that supports it. > It may be possible to send a reinvite with autoanswer headers but I > doubt that would work, all you could do is try making code to do it it > a sipp or sipsak scenario and test it. A better aproach might be to > answer the call normally and detect that to start your web workflow or > not really ring the phone, just the web app and deliver the call with > autoanswer when the button is hit in the web ui. > > Mike > > On Jun 16, 2009, at 4:24 AM, Peter P GMX wrote: > > >> Hello Brian, >> >> this is too easy :-). >> >> This is for a small callcenter app and I only want the user to pickup >> the call once (to accept the call in X-Lite (or a Snom phone) and to >> start the workflow on the web application). I do not want him to >> accept >> the call on the phone and then on the Web app. >> >> Best regards >> Peter >> >> >> >> Brian West schrieb: >> >>> click on the AA button? :) >>> >>> /b >>> >>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>> >>> >>> >>>> What is the best way to have this done? Move the call to park and >>>> then >>>> retransfer again with intercom, or is there a better solution? >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dujinfang at gmail.com Tue Jun 16 05:40:00 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 16 Jun 2009 20:40:00 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> Message-ID: <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 637 at 192.168.1.16 Contact: "user" Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. On Jun 16, 2009, at 7:35 PM, Michael Jerris wrote: > If you can catch brian or me on irc can you provide remote access to > this box and we should be able to fix this pretty quick > > Mike > > On Jun 16, 2009, at 5:20 AM, seven wrote: > >> Hi brain, >> >> Are you still looking into this? >> >> I think it must be some error when it register, I manually changed >> the contract str in the registration db, immediately it works. >> After re-register, stop work again. >> >> Should I report this to jira? >> >> sqlite> select contact from sip_registrations where contact like >> '%637%'; >> contact >> "user" > > >> >> sqlite> update sip_registrations set contact='"user" > >' where contact like '%637%'; >> >> >> >> >> On Jun 15, 2009, at 10:21 PM, Brian West wrote: >>> To: <"user" >>> >>> Can you reproduce this or let us in your box to look at it... >>> someone else reported this but I have yet to be able to reproduce >>> it. >>> >>> /b >>> >>> On Jun 15, 2009, at 2:41 AM, seven wrote: >>> >>>> Hi, >>>> >>>> I'm on version 13524, call from zoiper is ok, but when call >>>> zoiper, it keep rejecting calls, anyone can help? I'm seems >>>> always not the right time join in IRC :( >>>> >>>> http://pastebin.freeswitch.org/9383 >>>> >>>> >>>> Thanks. >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/715283ea/attachment-0001.html From brian at freeswitch.org Tue Jun 16 05:41:04 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 07:41:04 -0500 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A378D90.2040209@gmail.com> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> Message-ID: Can you please put it back to auto-nat and email me the output of global_getvar from the CLI so I can see what it detected? /b On Jun 16, 2009, at 7:18 AM, paul.degt at gmail.com wrote: > Solved by replacing "auto-nat" with public ip in public profile > "external_sip-ip" and "extrenal-rtp-ip" params. > I believe values for these params used to be taken from vars.xml and > so > would have public ips by default - would be nice to document such > changes in README. From brian at freeswitch.org Tue Jun 16 05:43:24 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 07:43:24 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37923D.8060809@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> Message-ID: Why not just keep the agent off hook.. in park state... then just playback ringing before you bridge? /b On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: > Hello Michael, > > I want the phone be ringing, just for acoustical feedback reasons. > > But what if I > > * transfer it to the same user destination again (now with intercom > enabled), will this work? > * transfer it to park and then transfer it to the same destination > again (now with intercom enabled) > > Best regards > Peter From brian at freeswitch.org Tue Jun 16 05:43:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 07:43:53 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> Message-ID: <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: > Almost caught you on IRC Mike. > > Our server is in a NAT'd network and all agents registered in the > same LAN. I can remotely register by using the public IP and the > contact string is right. > > Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. > User: 637 at 192.168.1.16 > Contact: "user" > > Agent: Zoiper rev.1809 > > So it's like only happens on our LAN and where there's a fs_path > present. > > Just curious, why agents registered on a local LAN has param > fs_nat=yes; (default internal profile, port 5060) ? > > Seems our time doesn't match, I'm generally available in office > 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/84dc0224/attachment.html From asannucci at gmail.com Tue Jun 16 05:51:12 2009 From: asannucci at gmail.com (bakko) Date: Tue, 16 Jun 2009 14:51:12 +0200 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: <24045890.post@talk.nabble.com> References: <24045890.post@talk.nabble.com> Message-ID: Did you compiled freeswitch with this command? ./configure --enable-core-odbc-support makemake installRegards From dujinfang at gmail.com Tue Jun 16 06:03:40 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 16 Jun 2009 21:03:40 +0800 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: References: <24045890.post@talk.nabble.com> Message-ID: <575E18F3-9021-4482-8CB6-C6799C6EDB7E@gmail.com> current configure will automatically use odbc if it's available, no need the --enable-core-odbc-support anymore. better to check if unixodbc-dev package installed of not. On Jun 16, 2009, at 8:51 PM, bakko wrote: > Did you compiled freeswitch with this command? > > ./configure --enable-core-odbc-support > makemake installRegards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Tue Jun 16 06:07:21 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 15:07:21 +0200 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> Message-ID: <4A379909.9050400@gmx.net> May this help also: I just tried current Zoiper with TLS. Outbound is working, inbound not. Zoiper registeres with the following contact info: "7233213" When a call comes in, Zoiper rings once and then hangs up. It shows "service or option not implemented" in the Zoiper log. My snom phones with the same parameters in the same network (they are all nated) register differently "723323" My FS logs show for an incoming call to Zoiper: 7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) Running State Change CS_CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) State CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [calling][0] 2009-06-16 14:50:16.340881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [terminated][415] 2009-06-16 14:50:16.340881 [NOTICE] sofia.c:3660 Hangup sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Its seems that something with the codecs fails here, although I have enabled all codecs in Zoiper and FS offers alaw. Best regards Peter Brian West schrieb: > Ok i'll have to se what I can do about reproducing this issue now that > I have more info on how its happening. > > /b > > On Jun 16, 2009, at 7:40 AM, dujinfang wrote: > >> Almost caught you on IRC Mike. >> >> Our server is in a NAT'd network and all agents registered in the >> same LAN. I can remotely register by using the public IP and the >> contact string is right. >> >> Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. >> User: 637 at 192.168.1.16 >> Contact: "user" >> >> Agent: Zoiper rev.1809 >> >> So it's like only happens on our LAN and where there's a fs_path present. >> >> Just curious, why agents registered on a local LAN has param >> fs_nat=yes; (default internal profile, port 5060) ? >> >> Seems our time doesn't match, I'm generally available in office >> 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. >> >> Thank you. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Tue Jun 16 06:17:25 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 16 Jun 2009 21:17:25 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> Message-ID: <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> What's wrong of the contact string? 639(snom) works but 637(zoiper) doesn't. "user" "seven" On Jun 16, 2009, at 8:43 PM, Brian West wrote: > Ok i'll have to se what I can do about reproducing this issue now > that I have more info on how its happening. > > /b > > On Jun 16, 2009, at 7:40 AM, dujinfang wrote: > >> Almost caught you on IRC Mike. >> >> Our server is in a NAT'd network and all agents registered in the >> same LAN. I can remotely register by using the public IP and the >> contact string is right. >> >> Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. >> User: 637 at 192.168.1.16 >> Contact: "user" > > >> Agent: Zoiper rev.1809 >> >> So it's like only happens on our LAN and where there's a fs_path >> present. >> >> Just curious, why agents registered on a local LAN has param >> fs_nat=yes; (default internal profile, port 5060) ? >> >> Seems our time doesn't match, I'm generally available in office >> 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. >> >> Thank you. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/cb7ae392/attachment.html From mike at jerris.com Tue Jun 16 06:21:42 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 08:21:42 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37923D.8060809@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> Message-ID: <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: > Hello Michael, > > I want the phone be ringing, just for acoustical feedback reasons. > > But what if I > > * transfer it to the same user destination again (now with intercom > enabled), will this work? > * transfer it to park and then transfer it to the same destination > again (now with intercom enabled) > > Best regards > Peter > > Michael Jerris schrieb: >> The only way I can think to do this today would be to cancel the call >> and re send with the intercom headers for a phone that supports it. >> It may be possible to send a reinvite with autoanswer headers but I >> doubt that would work, all you could do is try making code to do it >> it >> a sipp or sipsak scenario and test it. A better aproach might be to >> answer the call normally and detect that to start your web workflow >> or >> not really ring the phone, just the web app and deliver the call with >> autoanswer when the button is hit in the web ui. >> >> Mike >> >> On Jun 16, 2009, at 4:24 AM, Peter P GMX >> wrote: >> >> >>> Hello Brian, >>> >>> this is too easy :-). >>> >>> This is for a small callcenter app and I only want the user to >>> pickup >>> the call once (to accept the call in X-Lite (or a Snom phone) and to >>> start the workflow on the web application). I do not want him to >>> accept >>> the call on the phone and then on the Web app. >>> >>> Best regards >>> Peter >>> >>> >>> >>> Brian West schrieb: >>> >>>> click on the AA button? :) >>>> >>>> /b >>>> >>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>> >>>> >>>> >>>>> What is the best way to have this done? Move the call to park and >>>>> then >>>>> retransfer again with intercom, or is there a better solution? >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Jun 16 06:25:37 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 08:25:37 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> Message-ID: <9EB1D140-2A4E-4CA5-BE6E-458C1C436087@freeswitch.org> I need sip traces... also can you guys register to my dev box? dev.bkw.org with default user/pass try 1009 thru 1015 please. /b On Jun 16, 2009, at 8:17 AM, Seven Du wrote: > What's wrong of the contact string? 639(snom) works but 637(zoiper) > doesn't. > > "user" > > > "seven" > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/6850d80f/attachment.html From marketing at cluecon.com Tue Jun 16 07:24:20 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 16 Jun 2009 07:24:20 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Volunteers needed! Message-ID: <87f2f3b90906160724m159a3c36t2c840e9305ad59d7@mail.gmail.com> Spread the word! We have need of some volunteers to assist us with various tasks at ClueCon this year. As you may know, when putting on a conference there are numerous little things that require attention. Having several designated volunteers to handle these tasks will make the conference better for everyone. If you or someone you know would like to help out then please email me off list. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/3ca59299/attachment.html From panselva at gmail.com Tue Jun 16 08:17:59 2009 From: panselva at gmail.com (selva kumar) Date: Tue, 16 Jun 2009 20:47:59 +0530 Subject: [Freeswitch-users] Receiving calls us FS (Inbound) Message-ID: <45f609f90906160817i344c74c8i586687d65586f740@mail.gmail.com> Hi, I've tried configuring the inbound settings in default.xml, internal.xml, public.xml and acl.conf.xml. I am trying to route the call to one of the extension let's say 1005. It works well now. However, the outgoing is not happening but it worked find before Inbound is done. Now, when I remove the settings whatever I made to achieve inbound routing, the outbound works well. I am wondering like what needs to be made to achieve to blended environment. i.e. I need to be able to make outbound call and receive incoming calls. Request you to assist me in resolving the problem. Thanks Sam. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/92a5ce67/attachment.html From mike at jerris.com Tue Jun 16 09:14:58 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 12:14:58 -0400 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> Message-ID: This issue is now fixed in svn. Thanks Seven for access to your box to troubleshoot. Mike On Jun 16, 2009, at 9:17 AM, Seven Du wrote: > What's wrong of the contact string? 639(snom) works but 637(zoiper) > doesn't. > > "user" > > > "seven" > > > > On Jun 16, 2009, at 8:43 PM, Brian West wrote: >> Ok i'll have to se what I can do about reproducing this issue now >> that I have more info on how its happening. >> >> /b >> >> On Jun 16, 2009, at 7:40 AM, dujinfang wrote: >> >>> Almost caught you on IRC Mike. >>> >>> Our server is in a NAT'd network and all agents registered in the >>> same LAN. I can remotely register by using the public IP and the >>> contact string is right. >>> >>> Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. >>> User: 637 at 192.168.1.16 >>> Contact: "user" >> > >>> Agent: Zoiper rev.1809 >>> >>> So it's like only happens on our LAN and where there's a fs_path >>> present. >>> >>> Just curious, why agents registered on a local LAN has param >>> fs_nat=yes; (default internal profile, port 5060) ? >>> >>> Seems our time doesn't match, I'm generally available in office >>> 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. >>> >>> Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/7aca339a/attachment.html From raul at etellicom.com Tue Jun 16 09:16:05 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 16 Jun 2009 13:16:05 -0300 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A36E52D.2010005@gmx.net> References: <4A36E52D.2010005@gmx.net> Message-ID: <1245168965.11865.35.camel@raul-laptop> I actually do that with our call center application. For all incoming calls, our IVR engine parks the call in a virtual extension and plays back prompts, advertisements, MOH, process digits, etc. When the queue management finds an available agent, it sends an event to the client application for that agent (with an optional screen-pop) where the agent can click "Answer Call" and then we transfer the call with the auto-answer header set on to the agent phone. You could take a similar approach, if you're worrying about only providing ring-back tone to the caller you can simply park the call and use the playback app to play a tone_stream until the agent clicks the web link, which will transfer the call from the parking extension to the agent with the auto-answer flag. I'm still willing to make some tests with REINVITE providing auto-answer headers, as suggested by Mike. That would provide a more generic way to answer calls programmatically when it's already ringing the endpoint. I just need to find some time to read the sofia code and figure out how to do that :) Regards, Raul On Tue, 2009-06-16 at 02:19 +0200, Peter P GMX wrote: > I have managed to have a realtme status of a phone on a web page with > event_socket and a push service to the web bowser. > > What I am now trying to do is roughly the following: > > * when a call comes in, a flashing banner appears on the web page > with an underlying link (this works so far) > * when the user klicks on this flashing banner, the external SIP UA > which is already ringing, shall pick up the call. > > I know that it's possible to autoanswer a call with the intercom > feature. Also the SIP client X-Lite which we use here is able to > autoanswer a call. > I however want to manually decide when the UA takes the call with the > following workflow: > > * X-Lite rings on incoming call > * user klicks on the flashing banner > * X-Lite takes the call > > What is the best way to have this done? Move the call to park and then > retransfer again with intercom, or is there a better solution? > > Best regards > Peter > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Richard.Lamkin at mettoni.com Tue Jun 16 09:41:04 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 16 Jun 2009 17:41:04 +0100 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to an in bound call ? In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138052FDFE3@nickel.mettonigroup.com> Brian, Thank you for putting me on the right track. I thought I would share my results so after a bit of trial and error testing I came up with the follow DP rule, which lives in dialplan/public/. When an incoming call arrives for DDI 012345678 it is ack'ed with a "180 Ringing" and then the call is held up while the rule goes to sleep. On sleep expiry the call is cleared (from Ron McLeod's comment). This means any incoming call that is not processed using an API method will be automatically cleared after 3 mins. This makes a nice neat way of holing incoming calls ringing. Best Regards Richard Lamkin Richard.lamkin at mettonigroup.com From: Brian West [mailto:brian at freeswitch.org] Sent: 15 June 2009 21:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How do I get a 180 ringing to be sent toaninbound call ? Survey says ... "execute the ring_ready application" /b On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote: Something to consider is how long will be PSTN allow the call to remain un-answered. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Lamkin Sent: Monday, June 15, 2009 11:28 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin Richard.lamkin at mettonigroup.com ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/d0571c9e/attachment-0001.html From d at d-man.org Tue Jun 16 09:54:08 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 16 Jun 2009 09:54:08 -0700 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: <24045890.post@talk.nabble.com> References: <24045890.post@talk.nabble.com> Message-ID: What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the real logs from FS's logs? The info below is not nearly detailed enough. -----Original Message----- From: Edmar Cruz [mailto:darklion11 at yahoo.com] Sent: Monday, June 15, 2009 6:44 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER = root PASS = password ..... I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 890p24045890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From d at d-man.org Tue Jun 16 09:59:28 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 16 Jun 2009 09:59:28 -0700 Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed In-Reply-To: <200906091126.03556.yivzhenko@mksat.net> References: <200906091126.03556.yivzhenko@mksat.net> Message-ID: That should not be the case - I will double check this. My apologies if I broke it. :-( Please file a bug on this so I don't forget. _____ From: Yuriy Ivzhenko [mailto:yivzhenko at mksat.net] Sent: Tuesday, June 09, 2009 1:26 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed Some time ago mod_nibblebill was set variable nibble_total_billed after hangup. But after last few updates of module this variable is no more sets. Somebody else have this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/0bb621ef/attachment.html From d at d-man.org Tue Jun 16 09:59:46 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 16 Jun 2009 09:59:46 -0700 Subject: [Freeswitch-users] mod_niible install problem In-Reply-To: References: Message-ID: This should be fixed in the latest build (thanks MikeJ) _____ From: ram [mailto:talk2ram at gmail.com] Sent: Tuesday, June 09, 2009 12:03 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_niible install problem Hi i have downloaded latest SVN and trying to make install i get the following error I googled for the same but there no information on this error how can i resolve this problem Ram making install mod_nibblebill Compiling mod_nibblebill.c... Compiling mod_nibblebill.c ... mod_nibblebill.c: In function ?get_balance?: mod_nibblebill.c:368: error: ?balance? undeclared (first use in this function) mod_nibblebill.c:368: error: (Each undeclared identifier is reported only once mod_nibblebill.c:368: error: for each function it appears in.) make[5]: *** [mod_nibblebill.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_nibblebill-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/1e6c3979/attachment.html From msc at freeswitch.org Tue Jun 16 10:24:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Jun 2009 10:24:11 -0700 Subject: [Freeswitch-users] Receiving calls us FS (Inbound) In-Reply-To: <45f609f90906160817i344c74c8i586687d65586f740@mail.gmail.com> References: <45f609f90906160817i344c74c8i586687d65586f740@mail.gmail.com> Message-ID: <87f2f3b90906161024j382f004et1e33c5611ac7fb4e@mail.gmail.com> I think this question might need to be backed up with some more information. I recommend you post your relevant configs to pastebin so that we can have a look. (pastebin.freeswitch.org) -MC On Tue, Jun 16, 2009 at 8:17 AM, selva kumar wrote: > Hi, > > I've tried configuring the inbound settings in default.xml, internal.xml, > public.xml and acl.conf.xml. > > I am trying to route the call to one of the extension let's say 1005. It > works well now. However, the outgoing is not happening but it worked find > before Inbound is done. > > Now, when I remove the settings whatever I made to achieve inbound routing, > the outbound works well. I am wondering like what needs to be made to > achieve to blended environment. i.e. I need to be able to make outbound call > and receive incoming calls. > > Request you to assist me in resolving the problem. > > Thanks > Sam. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/18ad6ef7/attachment.html From marketing at cluecon.com Tue Jun 16 10:37:23 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 16 Jun 2009 10:37:23 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Getting Ready! Message-ID: <87f2f3b90906161037s44218ecbie96ca23d2df4664b@mail.gmail.com> ClueCon 2009 is only seven weeks away! We are all looking forward to meeting together in Chicago. To make sure that everything goes as planned we would like to know how many people will be attending. If you have not already signed up for ClueCon 2009 please do so. Call 877.742.CLUE and Brian will get you registered. Also, sign up at www.cluecon.com so that you can get updates on speakers, schedules, and sponsors. If you have any questions at all please feel free to call or email us. We look forward to seeing you this August! -Michael Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/b3529a09/attachment.html From Prometheus001 at gmx.net Tue Jun 16 10:49:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 19:49:16 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> Message-ID: <4A37DB1C.8070706@gmx.net> It mainly works now by uuid_transfer the following way via event socket. uuid_setvar sip_invite_params intercom=true uuid_setvar sip_auto_answer true uuid_transfer 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: ;transport=tls;line=er6kxnib Max-Forwards: 68 From: "Peter FS" ;tag=9eQ8rjQy533HF To: Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: Call-Info: ;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: "Peter FS" ;tag=9eQ8rjQy533HF To: ;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: ;reg-id=1 WWW-Authenticate: Digest realm="sip2.mycompany.de", nonce="2ee26efe6ab27f88", algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: > The transfer should work but it sounds like offhook agents is what > your really trying to accomplish so I would go with brian's suggestion. > > > > On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: > > >> Hello Michael, >> >> I want the phone be ringing, just for acoustical feedback reasons. >> >> But what if I >> >> * transfer it to the same user destination again (now with intercom >> enabled), will this work? >> * transfer it to park and then transfer it to the same destination >> again (now with intercom enabled) >> >> Best regards >> Peter >> >> Michael Jerris schrieb: >> >>> The only way I can think to do this today would be to cancel the call >>> and re send with the intercom headers for a phone that supports it. >>> It may be possible to send a reinvite with autoanswer headers but I >>> doubt that would work, all you could do is try making code to do it >>> it >>> a sipp or sipsak scenario and test it. A better aproach might be to >>> answer the call normally and detect that to start your web workflow >>> or >>> not really ring the phone, just the web app and deliver the call with >>> autoanswer when the button is hit in the web ui. >>> >>> Mike >>> >>> On Jun 16, 2009, at 4:24 AM, Peter P GMX >>> wrote: >>> >>> >>> >>>> Hello Brian, >>>> >>>> this is too easy :-). >>>> >>>> This is for a small callcenter app and I only want the user to >>>> pickup >>>> the call once (to accept the call in X-Lite (or a Snom phone) and to >>>> start the workflow on the web application). I do not want him to >>>> accept >>>> the call on the phone and then on the Web app. >>>> >>>> Best regards >>>> Peter >>>> >>>> >>>> >>>> Brian West schrieb: >>>> >>>> >>>>> click on the AA button? :) >>>>> >>>>> /b >>>>> >>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> What is the best way to have this done? Move the call to park and >>>>>> then >>>>>> retransfer again with intercom, or is there a better solution? >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Tue Jun 16 11:17:01 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 14:17:01 -0400 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37DB1C.8070706@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> <4A37DB1C.8070706@gmx.net> Message-ID: <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> uuid_setvar sip_invite_params intercom=true should be unnecessary. Mike On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: > It mainly works now by uuid_transfer the following way via event > socket. > uuid_setvar sip_invite_params intercom=true > uuid_setvar sip_auto_answer true > uuid_transfer 1000 XML default > so the call is transferred from 1000 to 1000. > > What happens: > 1) If I disable intercom on the Snom phone, the phone rings, stops > ringing and rings again (ok) > 1) If I enable intercom on the Snom phone, the phone rings, stops > ringing and hangs up (not ok) > > So I do not get the Snom to pick up the call in intercom mode. > > The last invite is: > INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib > SIP/2.0 > Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > Route: ;transport=tls;line=er6kxnib > Max-Forwards: 68 > From: "Peter FS" ;tag=9eQ8rjQy533HF > To: > > > Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > CSeq: 116467629 INVITE > Contact: > Call-Info: ;answer-after=0 > The intercom part is there and the Call-Info line with answer-after > also. > > The phone answers with > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > From: "Peter FS" ;tag=9eQ8rjQy533HF > To: > >;tag=71rskygkr2 > Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > CSeq: 116467629 INVITE > Contact: > ;reg-id=1 > WWW-Authenticate: Digest realm="sip2.mycompany.de", > nonce="2ee26efe6ab27f88", algorithm=MD5 > Content-Length: 0 > and hangs up. > > Anybody know how to solve this Snom intercom issue? > > Best regards > Peter > > > Michael Jerris schrieb: >> The transfer should work but it sounds like offhook agents is what >> your really trying to accomplish so I would go with brian's >> suggestion. >> >> >> >> On Jun 16, 2009, at 7:38 AM, Peter P GMX >> wrote: >> >> >>> Hello Michael, >>> >>> I want the phone be ringing, just for acoustical feedback reasons. >>> >>> But what if I >>> >>> * transfer it to the same user destination again (now with >>> intercom >>> enabled), will this work? >>> * transfer it to park and then transfer it to the same destination >>> again (now with intercom enabled) >>> >>> Best regards >>> Peter >>> >>> Michael Jerris schrieb: >>> >>>> The only way I can think to do this today would be to cancel the >>>> call >>>> and re send with the intercom headers for a phone that supports it. >>>> It may be possible to send a reinvite with autoanswer headers but I >>>> doubt that would work, all you could do is try making code to do it >>>> it >>>> a sipp or sipsak scenario and test it. A better aproach might be >>>> to >>>> answer the call normally and detect that to start your web workflow >>>> or >>>> not really ring the phone, just the web app and deliver the call >>>> with >>>> autoanswer when the button is hit in the web ui. >>>> >>>> Mike >>>> >>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX >>>> wrote: >>>> >>>> >>>> >>>>> Hello Brian, >>>>> >>>>> this is too easy :-). >>>>> >>>>> This is for a small callcenter app and I only want the user to >>>>> pickup >>>>> the call once (to accept the call in X-Lite (or a Snom phone) >>>>> and to >>>>> start the workflow on the web application). I do not want him to >>>>> accept >>>>> the call on the phone and then on the Web app. >>>>> >>>>> Best regards >>>>> Peter >>>>> >>>>> >>>>> >>>>> Brian West schrieb: >>>>> >>>>> >>>>>> click on the AA button? :) >>>>>> >>>>>> /b >>>>>> >>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> What is the best way to have this done? Move the call to park >>>>>>> and >>>>>>> then >>>>>>> retransfer again with intercom, or is there a better solution? >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bjbrashier at gmail.com Tue Jun 16 10:51:57 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 10:51:57 -0700 Subject: [Freeswitch-users] Voice lag in conference Message-ID: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/46b44ff5/attachment.html From msc at freeswitch.org Tue Jun 16 11:35:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Jun 2009 11:35:52 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> Message-ID: <87f2f3b90906161135r195d5d7ch491e54440f78964e@mail.gmail.com> Can you describe your networking environment a bit? One thing that can affect the latency of your voice traffic is your network infrastructure. If you can isolate FS and some phones on a separate, controlled network then possibly you can start narrowing it down to other factors. -MC On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier wrote: > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/d443c49c/attachment.html From anthony.minessale at gmail.com Tue Jun 16 11:47:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Jun 2009 13:47:41 -0500 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> Message-ID: <191c3a030906161147l177abfbfq804ed5fb71a77afd@mail.gmail.com> The problem comes from the timing of certain phones during the capture of audio actually clocked slightly faster than what it advertises. Try the latest trunk with all the defaults in your sip profile as we have tried to make the defaults deal with this automatically. On Tue, Jun 16, 2009 at 12:51 PM, Bradley Brashier wrote: > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/9be60aa8/attachment.html From bjbrashier at gmail.com Tue Jun 16 12:02:03 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 12:02:03 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <87f2f3b90906161135r195d5d7ch491e54440f78964e@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <87f2f3b90906161135r195d5d7ch491e54440f78964e@mail.gmail.com> Message-ID: <7bcfdd290906161202h78379859r9484a7aeed6e9877@mail.gmail.com> I have two different network setups, and have seen similar lag on both. The first is my home testbed. I'm connected to the internet through a home router and then a cablemodem. The home environment is pretty spare, of course. 2 machines and a couple of T-mobile cell phones with their SIP communication is all that goes through there. I have used my cell phone, a couple of different softphones, Gizmo call-ins, and regular PSTN calls. The worst lag is the T-mobile cell phones, but I'm happy to write that off as T-mobile's problem if we'd like. The second is the debug server environment on the systems where the conference product will eventually reside. The system is very complex, as it is already running a major hosted PBX service written years ago. I'm afraid all of the details of this system are beyond me, but I know that it includes a PSTN gateway, more T1s than I can count, and I'm having to split the RTP and SIP packets on separate ports for security and organizational purposes. For call-ins, I have used T-mobile again and regular PSTN, no softphones (yet). Obviously, this is the important environment, and the PSTN lag is somewhere around 500-700 ms (subjective). So am I correct in understanding that this is not a common issue, then, and that something can theoretically be done to help it? On Tue, Jun 16, 2009 at 11:35 AM, Michael Collins wrote: > Can you describe your networking environment a bit? One thing that can > affect the latency of your voice traffic is your network infrastructure. If > you can isolate FS and some phones on a separate, controlled network then > possibly you can start narrowing it down to other factors. > > -MC > > On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier > wrote: > >> I'm creating a conferencing product for use in a system with >> theoretically several hundred concurrent calls. I'm using FreeSwitch to >> create this product, but am not only new to FreeSwitch, but also the entire >> telecom industry as well as Open Source projects in general (I'm a >> recovering BIOS guy). >> >> I've got a bare-bones conference up and running on the server, including a >> handshake and a couple of features, and am using the default packages from >> the current trunk, but I've noticed that voice lag is a pretty big issue. >> Common lag times are several hundred milliseconds, and I've heard as long as >> a second. It seems to be at least marginally specific to individual phones >> -- certain phones have longer lag than others even on the same call. >> >> My question is really about what my options are. Is this just a part of >> SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim >> down that will help? Is this a common issue? If it's common, is it expected >> by the marketplace? >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/8064790e/attachment.html From joshm at wabashcenter.com Tue Jun 16 12:01:13 2009 From: joshm at wabashcenter.com (Josh Moon) Date: Tue, 16 Jun 2009 15:01:13 -0400 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> Message-ID: <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn't a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/c8ce5a1c/attachment-0001.html From Prometheus001 at gmx.net Tue Jun 16 13:11:05 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 22:11:05 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> <4A37DB1C.8070706@gmx.net> <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> Message-ID: <4A37FC59.60401@gmx.net> Thanks Michael, I have disabled it now. I finally got it to work, (sip_h_Call-Info=;answer-after=0) but the behaviour was not as desired, as I didn't manage the phone to pick up the call on the headset. It will only have the speaker enabled. So I will have to go a different way with parking the call and then forward it. Best regards Peter Michael Jerris schrieb: > uuid_setvar sip_invite_params intercom=true should be > unnecessary. > > Mike > > On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: > > >> It mainly works now by uuid_transfer the following way via event >> socket. >> uuid_setvar sip_invite_params intercom=true >> uuid_setvar sip_auto_answer true >> uuid_transfer 1000 XML default >> so the call is transferred from 1000 to 1000. >> >> What happens: >> 1) If I disable intercom on the Snom phone, the phone rings, stops >> ringing and rings again (ok) >> 1) If I enable intercom on the Snom phone, the phone rings, stops >> ringing and hangs up (not ok) >> >> So I do not get the Snom to pick up the call in intercom mode. >> >> The last invite is: >> INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib >> SIP/2.0 >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF >> Route: ;transport=tls;line=er6kxnib >> Max-Forwards: 68 >> From: "Peter FS" ;tag=9eQ8rjQy533HF >> To: >> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e >> CSeq: 116467629 INVITE >> Contact: >> Call-Info: ;answer-after=0 >> The intercom part is there and the Call-Info line with answer-after >> also. >> >> The phone answers with >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF >> From: "Peter FS" ;tag=9eQ8rjQy533HF >> To: >> > >>> ;tag=71rskygkr2 >>> >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e >> CSeq: 116467629 INVITE >> Contact: >> ;reg-id=1 >> WWW-Authenticate: Digest realm="sip2.mycompany.de", >> nonce="2ee26efe6ab27f88", algorithm=MD5 >> Content-Length: 0 >> and hangs up. >> >> Anybody know how to solve this Snom intercom issue? >> >> Best regards >> Peter >> >> >> Michael Jerris schrieb: >> >>> The transfer should work but it sounds like offhook agents is what >>> your really trying to accomplish so I would go with brian's >>> suggestion. >>> >>> >>> >>> On Jun 16, 2009, at 7:38 AM, Peter P GMX >>> wrote: >>> >>> >>> >>>> Hello Michael, >>>> >>>> I want the phone be ringing, just for acoustical feedback reasons. >>>> >>>> But what if I >>>> >>>> * transfer it to the same user destination again (now with >>>> intercom >>>> enabled), will this work? >>>> * transfer it to park and then transfer it to the same destination >>>> again (now with intercom enabled) >>>> >>>> Best regards >>>> Peter >>>> >>>> Michael Jerris schrieb: >>>> >>>> >>>>> The only way I can think to do this today would be to cancel the >>>>> call >>>>> and re send with the intercom headers for a phone that supports it. >>>>> It may be possible to send a reinvite with autoanswer headers but I >>>>> doubt that would work, all you could do is try making code to do it >>>>> it >>>>> a sipp or sipsak scenario and test it. A better aproach might be >>>>> to >>>>> answer the call normally and detect that to start your web workflow >>>>> or >>>>> not really ring the phone, just the web app and deliver the call >>>>> with >>>>> autoanswer when the button is hit in the web ui. >>>>> >>>>> Mike >>>>> >>>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX >>>>> wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Hello Brian, >>>>>> >>>>>> this is too easy :-). >>>>>> >>>>>> This is for a small callcenter app and I only want the user to >>>>>> pickup >>>>>> the call once (to accept the call in X-Lite (or a Snom phone) >>>>>> and to >>>>>> start the workflow on the web application). I do not want him to >>>>>> accept >>>>>> the call on the phone and then on the Web app. >>>>>> >>>>>> Best regards >>>>>> Peter >>>>>> >>>>>> >>>>>> >>>>>> Brian West schrieb: >>>>>> >>>>>> >>>>>> >>>>>>> click on the AA button? :) >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> What is the best way to have this done? Move the call to park >>>>>>>> and >>>>>>>> then >>>>>>>> retransfer again with intercom, or is there a better solution? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From bjbrashier at gmail.com Tue Jun 16 14:02:21 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 14:02:21 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> Message-ID: <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon wrote: > I am not as knowledgeable as the developers that will respond to your > question but I had the same problem as you. Here is what I did to combat > the delay: > > > > First off I started everything from scratch. I reinstalled Linux and then > I reinstalled FreeSWITCH by creating .deb packages. > > I then created my own conference profile and set the sample rate to 4000 > and changed the energy level to 20. > > I also made sure to test the conference room from phones that were in > completely different areas so there wasn?t a chance for feedback or really > bad echoing problems. > > > > Once I knew the delay was solved I raised the sample rate to 8000. I > tested it to make sure it would work properly. > > > > As Michael stated, this could be your network infrastructure but I just > wanted to let another FreeSWITCH user know what I did to try and stop the > voice delay. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley > Brashier > *Sent:* Tuesday, June 16, 2009 1:52 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Voice lag in conference > > > > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/35751cf0/attachment.html From bjbrashier at gmail.com Tue Jun 16 14:26:33 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 14:26:33 -0700 Subject: [Freeswitch-users] Controlling Conference Controls Message-ID: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing "lock" and "unlock" actions and merge them into a "lock toggle" action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? There are a variety of small things like this that I need to implement. Would I be better off switching to Lua? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/7fc87dcf/attachment.html From joshm at wabashcenter.com Tue Jun 16 14:30:56 2009 From: joshm at wabashcenter.com (Josh Moon) Date: Tue, 16 Jun 2009 17:30:56 -0400 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com><9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> Message-ID: <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> I was able to reduce it considerably. I can't say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 5:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Voice lag in conference I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon > wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn't a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/191d7951/attachment-0001.html From msc at freeswitch.org Tue Jun 16 14:38:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Jun 2009 14:38:42 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> Message-ID: <87f2f3b90906161438v2246837fme8068aa13ea51240@mail.gmail.com> What is the big picture application? Reason I ask is that the FS devs and community have a lot of experience so if they can see the big picture they might be able to offer better advice. -MC On Tue, Jun 16, 2009 at 2:26 PM, Bradley Brashier wrote: > How much power do I have with DTMF conference controls? The wiki doesn't > have much information on this. For example, one of the things I'd like to do > is take the currently existing "lock" and "unlock" actions and merge them > into a "lock toggle" action. Preferably in XML configuration files. Is this > even possible? If so, how would I get started? > > There are a variety of small things like this that I need to implement. > Would I be better off switching to Lua? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/4d45e82e/attachment.html From diego.viola at gmail.com Tue Jun 16 14:39:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 17:39:02 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? Message-ID: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego From intralanman at freeswitch.org Tue Jun 16 14:41:25 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 16 Jun 2009 17:41:25 -0400 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> Message-ID: <4A381185.9060806@freeswitch.org> Bradley Brashier wrote: > How much power do I have with DTMF conference controls? The wiki > doesn't have much information on this. For example, one of the things > I'd like to do is take the currently existing "lock" and "unlock" > actions and merge them into a "lock toggle" action. Preferably in XML > configuration files. Is this even possible? If so, how would I get > started? you could do this by having a script listen on the event socket... instead of using the default controls, you could just listen for a certain dtmf and then send the [un]lock command to the conference over the event socket -Ray From dule.maillist at gmail.com Tue Jun 16 14:43:28 2009 From: dule.maillist at gmail.com (Dan Le) Date: Tue, 16 Jun 2009 17:43:28 -0400 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24045824.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> Message-ID: <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> If you want FS server A to be able to call FS server B, you can set up a user account in server B's FS directory configs, and then just treat server B as a normal gateway by adding a gateway definition in server A. That will allow you to route calls to server B from A; to do the reverse, just mirror the configs the other direction. On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > > I like to connect two freeswitch, call each other, communicate and vice > versa. > Can you give me an example for that? > > Thanks > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/b2757e14/attachment.html From pete at privateconnect.com Tue Jun 16 14:48:40 2009 From: pete at privateconnect.com (pete at privateconnect.com) Date: Tue, 16 Jun 2009 14:48:40 -0700 Subject: [Freeswitch-users] =?utf-8?q?Which_GSM_gateway_to_buy=3F?= Message-ID: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/e698cd74/attachment.html From anthony.minessale at gmail.com Tue Jun 16 14:55:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Jun 2009 16:55:39 -0500 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> Message-ID: <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> don't forget to read my suggestion too from earlier today =D On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon wrote: > I was able to reduce it considerably. I can?t say it is completely gone > but I am very confident the ~.5 second delay I hear is because of the time > it takes my voice to go through the leaps and bounds of the phone company to > our server. I had at least a 3-5 second delay before I experimented with > the conference settings. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley > Brashier > *Sent:* Tuesday, June 16, 2009 5:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Voice lag in conference > > > > I'm not sure I've got the opportunity to do that at the moment, but I do > appreciate the point of view of a fellow product user. Were you able to > eliminate noticeable lag, or just reduce it to reasonable levels? > > > > I'll try to do something similar when I update to the newest trunk as > Anthony suggested. My copy is only a week old, but I'll try whatever has a > chance of working, and I know you guys have been working on conferencing > (the Moderator function couldn't have been timed better for me!). > > On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon > wrote: > > I am not as knowledgeable as the developers that will respond to your > question but I had the same problem as you. Here is what I did to combat > the delay: > > > > First off I started everything from scratch. I reinstalled Linux and then > I reinstalled FreeSWITCH by creating .deb packages. > > I then created my own conference profile and set the sample rate to 4000 > and changed the energy level to 20. > > I also made sure to test the conference room from phones that were in > completely different areas so there wasn?t a chance for feedback or really > bad echoing problems. > > > > Once I knew the delay was solved I raised the sample rate to 8000. I > tested it to make sure it would work properly. > > > > As Michael stated, this could be your network infrastructure but I just > wanted to let another FreeSWITCH user know what I did to try and stop the > voice delay. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley > Brashier > *Sent:* Tuesday, June 16, 2009 1:52 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Voice lag in conference > > > > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/df98338a/attachment-0001.html From bjbrashier at gmail.com Tue Jun 16 14:58:03 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 14:58:03 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> Message-ID: <7bcfdd290906161458n42e9479are572b462387f8adb@mail.gmail.com> Will do, just haven't had the time, yet! On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > don't forget to read my suggestion too from earlier today =D > > > > On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon wrote: > >> I was able to reduce it considerably. I can?t say it is completely gone >> but I am very confident the ~.5 second delay I hear is because of the time >> it takes my voice to go through the leaps and bounds of the phone company to >> our server. I had at least a 3-5 second delay before I experimented with >> the conference settings. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >> Brashier >> *Sent:* Tuesday, June 16, 2009 5:02 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Voice lag in conference >> >> >> >> I'm not sure I've got the opportunity to do that at the moment, but I do >> appreciate the point of view of a fellow product user. Were you able to >> eliminate noticeable lag, or just reduce it to reasonable levels? >> >> >> >> I'll try to do something similar when I update to the newest trunk as >> Anthony suggested. My copy is only a week old, but I'll try whatever has a >> chance of working, and I know you guys have been working on conferencing >> (the Moderator function couldn't have been timed better for me!). >> >> On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon >> wrote: >> >> I am not as knowledgeable as the developers that will respond to your >> question but I had the same problem as you. Here is what I did to combat >> the delay: >> >> >> >> First off I started everything from scratch. I reinstalled Linux and then >> I reinstalled FreeSWITCH by creating .deb packages. >> >> I then created my own conference profile and set the sample rate to 4000 >> and changed the energy level to 20. >> >> I also made sure to test the conference room from phones that were in >> completely different areas so there wasn?t a chance for feedback or really >> bad echoing problems. >> >> >> >> Once I knew the delay was solved I raised the sample rate to 8000. I >> tested it to make sure it would work properly. >> >> >> >> As Michael stated, this could be your network infrastructure but I just >> wanted to let another FreeSWITCH user know what I did to try and stop the >> voice delay. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >> Brashier >> *Sent:* Tuesday, June 16, 2009 1:52 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Voice lag in conference >> >> >> >> I'm creating a conferencing product for use in a system with theoretically >> several hundred concurrent calls. I'm using FreeSwitch to create this >> product, but am not only new to FreeSwitch, but also the entire telecom >> industry as well as Open Source projects in general (I'm a recovering BIOS >> guy). >> >> I've got a bare-bones conference up and running on the server, including a >> handshake and a couple of features, and am using the default packages from >> the current trunk, but I've noticed that voice lag is a pretty big issue. >> Common lag times are several hundred milliseconds, and I've heard as long as >> a second. It seems to be at least marginally specific to individual phones >> -- certain phones have longer lag than others even on the same call. >> >> My question is really about what my options are. Is this just a part of >> SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim >> down that will help? Is this a common issue? If it's common, is it expected >> by the marketplace? >> >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee you should not >> disseminate, distribute or copy this e-mail. Please notify the sender >> immediately by e-mail if you have received this e-mail by mistake and delete >> this e-mail from your system. E-mail transmission cannot be guaranteed to be >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender >> therefore does not accept liability for any errors or omissions in the >> contents of this message, which arise as a result of e-mail transmission. If >> verification is required please request a hard-copy version. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee you should not >> disseminate, distribute or copy this e-mail. Please notify the sender >> immediately by e-mail if you have received this e-mail by mistake and delete >> this e-mail from your system. E-mail transmission cannot be guaranteed to be >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender >> therefore does not accept liability for any errors or omissions in the >> contents of this message, which arise as a result of e-mail transmission. If >> verification is required please request a hard-copy version. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/3d05a52d/attachment.html From jmesquita at gmail.com Tue Jun 16 15:01:22 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 16 Jun 2009 19:01:22 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> Message-ID: <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> Get Khomp GSM cars! Ihihihih They will soon be compatible with FreeSWITCH. Laterz, jmesquita On Tue, Jun 16, 2009 at 6:48 PM, wrote: > I did a fair amount of research into GSM gateways about 8 months ago. I > should first ask what are you looking to do with the gateway? > > -pete > > > -------- Original Message -------- > Subject: [Freeswitch-users] Which GSM gateway to buy? > From: Diego Viola > Date: Tue, June 16, 2009 2:39 pm > To: freeswitch-users at lists.freeswitch.org > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/062a6b65/attachment.html From diego.viola at gmail.com Tue Jun 16 15:01:35 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 18:01:35 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> Message-ID: <86a32abc0906161501h2f5b634eq20de9eb2a58486ec@mail.gmail.com> I need it for gsm termination, I'd like to start with 8 channels, then 16, etc. Thanks, Diego On Tue, Jun 16, 2009 at 5:48 PM, wrote: > I did a fair amount of research into GSM gateways about 8 months ago.? I > should first ask what are you looking to do with the gateway? > > -pete > > -------- Original Message -------- > Subject: [Freeswitch-users] Which GSM gateway to buy? > From: Diego Viola > Date: Tue, June 16, 2009 2:39 pm > To: freeswitch-users at lists.freeswitch.org > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From edpimentl at gmail.com Tue Jun 16 15:09:35 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 16 Jun 2009 18:09:35 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> Message-ID: <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> For those that understand Portuguese http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a61e8051/attachment-0001.html From diego.viola at gmail.com Tue Jun 16 15:21:13 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 18:21:13 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> Message-ID: <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> We are a start-up company btw. On Tue, Jun 16, 2009 at 6:09 PM, EdPimentl wrote: > For those that understand Portuguese > http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve at justfone.com Tue Jun 16 15:21:06 2009 From: steve at justfone.com (Steven Brown) Date: Tue, 16 Jun 2009 23:21:06 +0100 Subject: [Freeswitch-users] Which GSM gateway to buy? Message-ID: <3e6d7b0c0906161521k204b9cawf19789d50fc92c98@mail.gmail.com> Hi , I have used PORTech single and double channel units on a couple of small projects with FS and they seem to have worked well in a low volume application . Have never tried one of the larger channel count ones yet for high call volumes though so cant verify how they perform, although just starting a larger project using 3 x 8 SIM PORTech units so will be able to give feedback on these in a few weeks. Steve Message: 2 Date: Tue, 16 Jun 2009 17:39:02 -0400 From: Diego Viola Subject: [Freeswitch-users] Which GSM gateway to buy? To: freeswitch-users at lists.freeswitch.org Message-ID: <86a32abc0906161439v89fbb58kcfe8297687dee600 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/b7524e13/attachment.html From diego.viola at gmail.com Tue Jun 16 15:22:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 18:22:47 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> Message-ID: <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> So we can't afford the top and the latest hardware. On Tue, Jun 16, 2009 at 6:21 PM, Diego Viola wrote: > We are a start-up company btw. > > On Tue, Jun 16, 2009 at 6:09 PM, EdPimentl wrote: >> For those that understand Portuguese >> http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ >> -E >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From paul.degt at gmail.com Tue Jun 16 15:50:51 2009 From: paul.degt at gmail.com (paul.degt) Date: Tue, 16 Jun 2009 18:50:51 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> Message-ID: <4A3821CB.2070904@gmail.com> API CALL [global_getvar()] output: external_ssl_enable=false external_tls_port=5081 external_sip_port=5080 external_auth_calls=false internal_ssl_dir=/var/opt/freeswitch/conf/ssl internal_sip_port=5060 default_provider_contact=5000 default_provider_from_domain=example.com default_provider_password=password external_rtp_ip=74.92.196.241 xmpp_server_profile=xmpps xmpp_client_profile=xmppc global_codec_prefs=G722,PCMU,PCMA,GSM hold_music=local_stream://moh external_ssl_dir=/var/opt/freeswitch/conf/ssl internal_auth_calls=true local_ip_v4=192.168.0.40 unroll_loops=true default_areacode=918 default_provider_register=false local_mask_v4=255.255.255.0 default_password=1234 call_debug=false local_ip_v6=::1 default_provider_username=joeuser sound_prefix=/var/opt/freeswitch/sounds/en/us/callie outbound_caller_id=0000000000 default_country=US base_dir=/var/opt/freeswitch bind_server_ip=auto internal_tls_port=5061 switch_serial=c0a8002854db default_provider=example.com outbound_codec_prefs=PCMU,PCMA,GSM domain_name=192.168.0.40 domain=192.168.0.40 external_sip_ip=74.92.196.241 outbound_caller_name=Versafon.com rs-ring=%(1000, 4000, 425.0, 0.0) sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) internal_ssl_enable=false console_loglevel=debug uk-ring=%(400,200,400,450);%(400,2200,400,450) us-ring=%(2000, 4000, 440.0, 480.0) sip_tls_version=tlsv1 fr-ring=%(1500, 3500, 440.0, 0.0) bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) Brian West wrote: > Can you please put it back to auto-nat and email me the output of > global_getvar from the CLI so I can see what it detected? > > /b > > On Jun 16, 2009, at 7:18 AM, paul.degt at gmail.com wrote: > > >> Solved by replacing "auto-nat" with public ip in public profile >> "external_sip-ip" and "extrenal-rtp-ip" params. >> I believe values for these params used to be taken from vars.xml and >> so >> would have public ips by default - would be nice to document such >> changes in README. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve at justfone.com Tue Jun 16 15:57:53 2009 From: steve at justfone.com (Steven Brown) Date: Tue, 16 Jun 2009 23:57:53 +0100 Subject: [Freeswitch-users] How to delay audio ? Message-ID: <3e6d7b0c0906161557w4091d00ck4169ab952a671d56@mail.gmail.com> Hi All, I have a requirement to delay the audio sent from the calling channel in a call by a specified delay, much the same as the delay_echo functionality in the dptools but in a bridged rather than loopback mode. I cant immediately see a way to achieve this, is this something I'm missing or should I have look at adapting the delay_echo functionality. Thanks Steve Steven Brown email steve at justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. From anthony.minessale at gmail.com Tue Jun 16 16:22:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Jun 2009 18:22:31 -0500 Subject: [Freeswitch-users] How to delay audio ? In-Reply-To: <3e6d7b0c0906161557w4091d00ck4169ab952a671d56@mail.gmail.com> References: <3e6d7b0c0906161557w4091d00ck4169ab952a671d56@mail.gmail.com> Message-ID: <191c3a030906161622k1b1444f6o350e043021539770@mail.gmail.com> if it's sip, turn on the jiterbuffer before you answer set the var jitterbuffer_msec=x where x is desired number of milliseconds (not too much!) On Tue, Jun 16, 2009 at 5:57 PM, Steven Brown wrote: > Hi All, > > I have a requirement to delay the audio sent from the calling channel > in a call by a specified delay, much the same as the delay_echo > functionality in the dptools but in a bridged rather than loopback > mode. I cant immediately see a way to achieve this, is this something > I'm missing or should I have look at adapting the delay_echo > functionality. > > Thanks > > Steve > > Steven Brown > > email steve at justfone.com > office 08707706968 > mobile 07768755409 > fax 07884636663 > > Justfone - Company Reg. No. : 3926817 > > Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 > 1EW > > The contents of this e-mail may be privileged and are confidential. It > may not be disclosed to or used by anyone other than the addressee(s), > nor copied in any way. If received in error, please advise sender, > then delete it from your system. Internet email communications are not > secure and therefore Justfone do not accept legal responsibility for > the contents of this message. Any views or opinions presented are > solely those of the author and do not necessarily represent those of > Justfone unless otherwise specifically stated. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/020f41fa/attachment.html From bjbrashier at gmail.com Tue Jun 16 16:33:22 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 16:33:22 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A381185.9060806@freeswitch.org> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> Message-ID: <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> Hmmm.... is that going to be easier than just modifying the mod_conference code to allow for a handfull of extra, simple commands? To me, it seems like for reasons of maintainability, etc, you want as few varied pieces as possible, in as few languages as possible. Socket scripting doesn't sound like it would be an extension of what I'm doing, now, more like a totally new method. Of course, I'm saying this from a complete outside point of view, and am more than willing to admit that I don't necessarily know the best course. On Tue, Jun 16, 2009 at 2:41 PM, Raymond Chandler < intralanman at freeswitch.org> wrote: > Bradley Brashier wrote: > > How much power do I have with DTMF conference controls? The wiki > > doesn't have much information on this. For example, one of the things > > I'd like to do is take the currently existing "lock" and "unlock" > > actions and merge them into a "lock toggle" action. Preferably in XML > > configuration files. Is this even possible? If so, how would I get > > started? > you could do this by having a script listen on the event socket... > instead of using the default controls, you could just listen for a > certain dtmf and then send the [un]lock command to the conference over > the event socket > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/4e83a6a2/attachment.html From william.suffill at gmail.com Tue Jun 16 17:04:36 2009 From: william.suffill at gmail.com (William Suffill) Date: Tue, 16 Jun 2009 20:04:36 -0400 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> Message-ID: <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> It depends pretty heavily on what you are trying to add function wise. If it's more in depth using the event socket would allow it to be used on any FreeSwitch server assuming it caught the dtmf and acted according without having to modify the core source code/recompile. It might be a bit more work at first but could be well worth it depending on your needs. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/786acbf3/attachment-0001.html From evilla at chipoly.com Tue Jun 16 18:22:48 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Tue, 16 Jun 2009 19:22:48 -0600 Subject: [Freeswitch-users] MadBoss Conferences Examples - bug? Message-ID: <000601c9eeea$20a56d00$61f04700$@com> Hello friends. I've been playing with the mad boss examples. There is an issue I'd like to see: For example in MadBoss3: The first leg added to conference is the loopback/9999. Then you can add more users by conference_set_auto_outcall function. The problem I see is that: 1) Loopback music is still in the background of conference. 2) When everyone hang up, the conference is still active, because the 9999 user (music) is still inside the room. How can music be stoped once meeting is going to start? Edwin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/f8290b42/attachment.html From darklion11 at yahoo.com Tue Jun 16 18:27:32 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:27:32 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> Message-ID: <24065535.post@talk.nabble.com> Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:28:21 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:28:21 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24065535.post@talk.nabble.com> Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:28:55 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:28:55 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24065535.post@talk.nabble.com> Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:42:15 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:42:15 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: References: <24045890.post@talk.nabble.com> Message-ID: <24065638.post@talk.nabble.com> my nibble.conf.xml Account 1001.xml I check unixodbc has been installed. # isql zenoss edmar edmar [SQL]> Connected successfully but on freeswitch error Cannot connect to user ODBC [root] Darren Schreiber wrote: > > What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the > real logs from FS's logs? The info below is not nearly detailed enough. > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Monday, June 15, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC > > > Hi > > I experiencing an error on mod_nibblebill. I already load it from > autoload_configs, especially mod_spidermonkey. Uncomment > mod_spidermonkey_odbc. I also download unixodbc and created the files > /etc/odbcinst.ini and /etc/odbc.ini with the correct format > > [zenoss] > DATABASE = tcapi > USER = root > PASS = password > ..... > > I type also on the console isql zenoss root password. Also working... > > But an error occur on freeswitch Cannot connect to user [root] ... > > What do you thinks is the problem? > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24045890.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24065638.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:43:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:43:59 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Message-ID: <24065638.post@talk.nabble.com> my nibble.conf.xml param name="db_username" value="edmar"/> param name="db_password" value="edmar"/> param name="db_dsn" value="tcapi"/> param name="db_column_cash" value="cash"/> param name="db_column_account" value="id"/> param name="global_heartbeat" value="1"/> !-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. --> param name="lowbal_amt" value="5"/> param name="lowbal_action" value="play ding"/> param name="nobal_amt" value="0"/> param name="nobal_action" value="hangup"/> param name="percall_max_amt" value="100"/> param name="percall_action" value="hangup"/> Account 1001.xml param name="password" value="1234"/> param name="vm-password" value="1001"/> param name="vm-mailto" value=""/> param name="vm-email-all-messages" value="false"/> param name="vm-delete-file" value="false"/> param name="vm-attach-file" value="false"/> I check unixodbc has been installed. # isql zenoss edmar edmar [SQL]> Connected successfully but on freeswitch error Cannot connect to user ODBC [root] Darren Schreiber wrote: > > What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the > real logs from FS's logs? The info below is not nearly detailed enough. > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Monday, June 15, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC > > > Hi > > I experiencing an error on mod_nibblebill. I already load it from > autoload_configs, especially mod_spidermonkey. Uncomment > mod_spidermonkey_odbc. I also download unixodbc and created the files > /etc/odbcinst.ini and /etc/odbc.ini with the correct format > > [zenoss] > DATABASE = tcapi > USER = root > PASS = password > ..... > > I type also on the console isql zenoss root password. Also working... > > But an error occur on freeswitch Cannot connect to user [root] ... > > What do you thinks is the problem? > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24045890.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24065638.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at gmail.com Tue Jun 16 18:45:07 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 16 Jun 2009 22:45:07 -0300 Subject: [Freeswitch-users] MadBoss Conferences Examples - bug? In-Reply-To: <000601c9eeea$20a56d00$61f04700$@com> References: <000601c9eeea$20a56d00$61f04700$@com> Message-ID: <5a8712120906161845t228d192eo80b52c38f65bdaf8@mail.gmail.com> Look at the newly implemented wait-mod conference flag on mod_conference. This is: http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E under parameters->conference-flags jmesquita On Tue, Jun 16, 2009 at 10:22 PM, Ing. Edwin Villarreal wrote: > Hello friends. > > > > I?ve been playing with the mad boss examples. There is an issue I?d like > to see: > > > > For example in MadBoss3: > > The first leg added to conference is the loopback/9999? Then you can add > more users by conference_set_auto_outcall function. > > > > The problem I see is that: > > 1) Loopback music is still in the background of conference. > > 2) When everyone hang up, the conference is still active, because the > 9999 user (music) is still inside the room. > > > > How can music be stoped once meeting is going to start? > > > > *Edwin* > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/911fff4c/attachment.html From cesar.bermudez at gmail.com Tue Jun 16 19:11:05 2009 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 16 Jun 2009 23:11:05 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> Message-ID: Diego, i'have a customer using 3 portech using todo termination on argentina with asterisk on high volume calls and they are working great. Best regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/c86d3f46/attachment.html From edpimentl at gmail.com Tue Jun 16 19:34:27 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 16 Jun 2009 22:34:27 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> Message-ID: <9dc4a1670906161934j24f2fcfej93d0286a0473fe0@mail.gmail.com> I have been using Portech for over two years and they work fine. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/f9d3a1a0/attachment.html From brian at freeswitch.org Tue Jun 16 19:49:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 21:49:58 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24065535.post@talk.nabble.com> References: <24065535.post@talk.nabble.com> Message-ID: Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > Is there another way to manage the gateway with the caller id of the > user > not the gateway user id and is there a gateway that doesn't need a > username > and password? From evilla at chipoly.com Tue Jun 16 19:58:20 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Tue, 16 Jun 2009 20:58:20 -0600 Subject: [Freeswitch-users] [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() Message-ID: <003501c9eef7$79330790$6b9916b0$@com> Hello! I need some fresh ideas about this issue. My gateway is already REGED, but when REG expires and sofia is trying to renew REG, then it fails to register. . 2009-06-16 16:46:39 [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() chipoly Registration Failed with status DNS Error [503]. failure #14 . 2009-06-16 16:46:40 [WARNING] sofia_reg.c:334 sofia_reg_check_gateway() chipoly Failed Registration, setting retry to 450 seconds. Here is a complete before/after http://pastebin.freeswitch.org/9406 when doing sofia profile external restart, gateway REGs again, so it's not DNS problem. (I think) Thank you for ur help! Edwin Villarreal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/333eeaa7/attachment.html From brian at freeswitch.org Tue Jun 16 20:03:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:03:05 -0500 Subject: [Freeswitch-users] [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() In-Reply-To: <003501c9eef7$79330790$6b9916b0$@com> References: <003501c9eef7$79330790$6b9916b0$@com> Message-ID: <16C354B9-9744-4BFB-8DDD-A793A96D9E1A@freeswitch.org> This should be a huge clue... what might be your providers name? Seems something is missing here or you have the settings wrong. /b On Jun 16, 2009, at 9:58 PM, Ing. Edwin Villarreal wrote: > DNS Error [503]. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/32af6f9c/attachment.html From darklion11 at yahoo.com Tue Jun 16 20:05:22 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 20:05:22 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> Message-ID: <24066210.post@talk.nabble.com> How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 On 192.168.0.4 Brian West-3 wrote: > > Turn off authentication or use ACL's > > /b > > On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > >> Is there another way to manage the gateway with the caller id of the >> user >> not the gateway user id and is there a gateway that doesn't need a >> username >> and password? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066210.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jonny.voip at gmail.com Tue Jun 16 20:05:54 2009 From: jonny.voip at gmail.com (Jonathan DiVita) Date: Tue, 16 Jun 2009 23:05:54 -0400 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 Message-ID: Hello, all. I'm currently playing around with a new install of Freeswitch and wanted to try out mod_opal. Below are the current SVN builds for opal, ptlib, and freeswitch. I end up with the following errors when compiling. making all mod_opal Compiling mod_opal.cpp... Compiling mod_opal.cpp ... In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ?virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)? /usr/include/opal/opal/localep.h:267: error: overriding ?virtual ptlib_virtual_function_changed_or_removed****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)?: mod_opal.cpp:564: error: no matching function for call to ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)? /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) make[4]: *** [mod_opal.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opal-all] Error 1 make[1]: *** [mod_opal] Error 2 make: *** [mod_opal] Error 2 root at Freeswitch1:~/opal# svn info Path: . URL: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/trunk Repository Root: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip Repository UUID: 023b2edf-31b2-4de3-b41e-bca80c47788f Revision: 22909 Node Kind: directory Schedule: normal Last Changed Author: rjongbloed Last Changed Rev: 22909 Last Changed Date: 2009-06-16 07:09:41 -0400 (Tue, 16 Jun 2009) root at Freeswitch1:~/opal# cd .. root at Freeswitch1:~# cd ptlib/ root at Freeswitch1:~/ptlib# svn info Path: . URL: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunk Repository Root: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip Repository UUID: 023b2edf-31b2-4de3-b41e-bca80c47788f Revision: 22909 Node Kind: directory Schedule: normal Last Changed Author: csoutheren Last Changed Rev: 22907 Last Changed Date: 2009-06-16 05:49:19 -0400 (Tue, 16 Jun 2009) root at Freeswitch1:~/ptlib# cd /freeswitch/ root at Freeswitch1:/freeswitch# svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 13798 Node Kind: directory Schedule: normal Last Changed Author: brian Last Changed Rev: 13798 Last Changed Date: 2009-06-16 19:11:45 -0400 (Tue, 16 Jun 2009) Do I need earlier versions of opal and ptlib? Thanks! Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/87ecc74a/attachment-0001.html From brian at freeswitch.org Tue Jun 16 20:12:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:12:01 -0500 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 In-Reply-To: References: Message-ID: please see MODOPAL-10 on jira. /b On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > Hello, all. I'm currently playing around with a new install of > Freeswitch and wanted to try out mod_opal. Below are the current > SVN builds for opal, ptlib, and freeswitch. I end up with the > following errors when compiling. > > making all mod_opal > Compiling mod_opal.cpp... > Compiling mod_opal.cpp ... > In file included from mod_opal.cpp:25: > mod_opal.h:151: error: conflicting return type specified for > ?virtual OpalLocalConnection* > FSEndPoint::CreateConnection(OpalCall&, void*)? > /usr/include/opal/opal/localep.h:267: error: overriding ?virtual > ptlib_virtual_function_changed_or_removed****** > OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? > mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, > FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, > switch_channel_t*)?: > mod_opal.cpp:564: error: no matching function for call to > ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, > NULL)? > /usr/include/opal/opal/localep.h:290: note: candidates are: > OpalLocalConnection::OpalLocalConnection(OpalCall&, > OpalLocalEndPoint&, void*, unsigned int, > OpalConnection::StringOptions*, char) > /usr/include/opal/opal/localep.h:276: note: > OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) > make[4]: *** [mod_opal.lo] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_opal-all] Error 1 > make[1]: *** [mod_opal] Error 2 > make: *** [mod_opal] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment.html From darklion11 at yahoo.com Tue Jun 16 20:03:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24065535.post@talk.nabble.com> Message-ID: <929908.28142.qm@web57310.mail.re1.yahoo.com> How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 On 192.168.0.4 ________________________________ From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 16, 2009 10:49:58 PM Subject: Re: [Freeswitch-users] How can I join two freeswitch on two servers? Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > Is there another way to manage the gateway with the caller id of the > user > not the gateway user id and is there a gateway that doesn't need a > username > and password? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/2839a86d/attachment.html From brian at freeswitch.org Tue Jun 16 20:31:21 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:31:21 -0500 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A3821CB.2070904@gmail.com> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> Message-ID: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> Shouldn't have really changed any behavior at all... What svn rev are you on? /b On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > API CALL [global_getvar()] output: > external_ssl_enable=false > external_tls_port=5081 > external_sip_port=5080 > external_auth_calls=false > internal_ssl_dir=/var/opt/freeswitch/conf/ssl > internal_sip_port=5060 > default_provider_contact=5000 > default_provider_from_domain=example.com > default_provider_password=password > external_rtp_ip=74.92.196.241 > xmpp_server_profile=xmpps > xmpp_client_profile=xmppc > global_codec_prefs=G722,PCMU,PCMA,GSM > hold_music=local_stream://moh > external_ssl_dir=/var/opt/freeswitch/conf/ssl > internal_auth_calls=true > local_ip_v4=192.168.0.40 > unroll_loops=true > default_areacode=918 > default_provider_register=false > local_mask_v4=255.255.255.0 > default_password=1234 > call_debug=false > local_ip_v6=::1 > default_provider_username=joeuser > sound_prefix=/var/opt/freeswitch/sounds/en/us/callie > outbound_caller_id=0000000000 > default_country=US > base_dir=/var/opt/freeswitch > bind_server_ip=auto > internal_tls_port=5061 > switch_serial=c0a8002854db > default_provider=example.com > outbound_codec_prefs=PCMU,PCMA,GSM > domain_name=192.168.0.40 > domain=192.168.0.40 > external_sip_ip=74.92.196.241 > outbound_caller_name=Versafon.com > rs-ring=%(1000, 4000, 425.0, 0.0) > sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) > internal_ssl_enable=false > console_loglevel=debug > uk-ring=%(400,200,400,450);%(400,2200,400,450) > us-ring=%(2000, 4000, 440.0, 480.0) > sip_tls_version=tlsv1 > fr-ring=%(1500, 3500, 440.0, 0.0) > bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) > From brian at freeswitch.org Tue Jun 16 20:36:38 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:36:38 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <929908.28142.qm@web57310.mail.re1.yahoo.com> References: <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> Message-ID: Now you have to tell the sofia profile to use that ACL /b On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > How can i turn off authentication? This is my acl.conf.xml on > 192.168.0.105 > > > > > > On 192.168.0.4 > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/47e21916/attachment.html From darklion11 at yahoo.com Tue Jun 16 21:00:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:00:04 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> Message-ID: <24066647.post@talk.nabble.com> How can sofia profile can call ACL? Can you give me an example? Brian West-3 wrote: > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 21:02:21 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:02:21 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24066647.post@talk.nabble.com> How can sofia profile can call ACL? Can you give me an example? Like this? I put this on external profile "/> "/> Brian West-3 wrote: > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 16 21:08:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 23:08:46 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066647.post@talk.nabble.com> References: <24066647.post@talk.nabble.com> Message-ID: <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> COPY paste fail :) something like that as per the example. /b On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > > How can sofia profile can call ACL? > Can you give me an example? > Like this? > > I put this on external profile > > "/> > "/> > > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From darklion11 at yahoo.com Tue Jun 16 21:26:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:26:38 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> Message-ID: <24066825.post@talk.nabble.com> If FS A has an account 8011105 does FS B also nid to register 8011105? Yes it working on a gateway but the username of the gateway was shown on my softphone and also it nids a password for the gateway... is there an option to view the caller name and number of the FS A gateway to FS B? Brian West-3 wrote: > > COPY paste fail :) > > > > something like that as per the example. > > /b > > On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > >> >> How can sofia profile can call ACL? >> Can you give me an example? >> Like this? >> >> I put this on external profile >> >> "/> >> "/> >> >> >> Brian West-3 wrote: >>> >>> Now you have to tell the sofia profile to use that ACL >>> >>> /b >>> >>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>> >>>> How can i turn off authentication? This is my acl.conf.xml on >>>> 192.168.0.105 >>>> >>>> >>>> >>>> >>>> >>>> On 192.168.0.4 >>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 21:30:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:30:38 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066825.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> Message-ID: <24066849.post@talk.nabble.com> Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on FS B Edmar Cruz wrote: > > If FS A has an account 8011105 does FS B also nid to register 8011105? Yes > it working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066849.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 16 21:33:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 23:33:56 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066849.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24066849.post@talk.nabble.com> Message-ID: Its not an error its a warning and you don't have your ACL's configured correctly. You're trying too hard! :) set auth- calls=false on the profile. /b On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote: > > Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on > FS B From bipin at xbipin.com Tue Jun 16 22:00:59 2009 From: bipin at xbipin.com (xbipin) Date: Tue, 16 Jun 2009 22:00:59 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <86a32abc0906160301p802a71arb8d67bc9a0ebcfa9@mail.gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> <24050713.post@talk.nabble.com> <86a32abc0906160301p802a71arb8d67bc9a0ebcfa9@mail.gmail.com> Message-ID: <24067052.post@talk.nabble.com> hi, if u need any help, i can always provide that. Regards, Bipin Diego Viola wrote: > > Sure, I will let you know when it's done. > > On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruz wrote: >> >> Thanks for that info... Can you send me this project if and only if it is >> already finished on this email darklion at yahoo.com? Thanks a lot... >> >> >> Diego Viola wrote: >>> >>> I'm currently rewriting the entire thing, it was a commercial app >>> first, but I'm re-writing it in order to make it open source. It's not >>> ready yet, as soon as I finish it, I will release it to the public. >>> >>> Diego >>> >>> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz >>> wrote: >>>> >>>> Can you share me the link of it so i can try... Please >>>> >>>> Diego Viola wrote: >>>>> >>>>> I'm currently writing a rails app that uses mod_nibblebill for >>>>> billing, >>>>> it's >>>>> a calling card app. >>>>> >>>>> Diego >>>>> >>>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>>>> wrote: >>>>> >>>>>> >>>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>>>> fully >>>>>> developed... >>>>>> Is there any GUI with billing options? >>>>>> >>>>>> >>>>>> seven-8 wrote: >>>>>> > >>>>>> > http://www.tcapi.org/index.php?title=Main_Page >>>>>> > >>>>>> > >>>>>> > >>>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>>>> > >>>>>> >> >>>>>> >> is there any available gui for freeswitch using cake php complete >>>>>> >> instead of >>>>>> >> wikipbx, spice softphone or pfsense? >>>>>> >> -- >>>>>> >> View this message in context: >>>>>> >> >>>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>> >> >>>>>> >> >>>>>> >> _______________________________________________ >>>>>> >> Freeswitch-users mailing list >>>>>> >> Freeswitch-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Freeswitch-users mailing list >>>>>> > Freeswitch-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24067052.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From paul.degt at gmail.com Tue Jun 16 22:00:36 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Wed, 17 Jun 2009 01:00:36 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> Message-ID: <4A387874.5090105@gmail.com> 13564 Brian West wrote: > Shouldn't have really changed any behavior at all... What svn rev are > you on? > > /b > > On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > > >> API CALL [global_getvar()] output: >> external_ssl_enable=false >> external_tls_port=5081 >> external_sip_port=5080 >> external_auth_calls=false >> internal_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_sip_port=5060 >> default_provider_contact=5000 >> default_provider_from_domain=example.com >> default_provider_password=password >> external_rtp_ip=74.92.196.241 >> xmpp_server_profile=xmpps >> xmpp_client_profile=xmppc >> global_codec_prefs=G722,PCMU,PCMA,GSM >> hold_music=local_stream://moh >> external_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_auth_calls=true >> local_ip_v4=192.168.0.40 >> unroll_loops=true >> default_areacode=918 >> default_provider_register=false >> local_mask_v4=255.255.255.0 >> default_password=1234 >> call_debug=false >> local_ip_v6=::1 >> default_provider_username=joeuser >> sound_prefix=/var/opt/freeswitch/sounds/en/us/callie >> outbound_caller_id=0000000000 >> default_country=US >> base_dir=/var/opt/freeswitch >> bind_server_ip=auto >> internal_tls_port=5061 >> switch_serial=c0a8002854db >> default_provider=example.com >> outbound_codec_prefs=PCMU,PCMA,GSM >> domain_name=192.168.0.40 >> domain=192.168.0.40 >> external_sip_ip=74.92.196.241 >> outbound_caller_name=Versafon.com >> rs-ring=%(1000, 4000, 425.0, 0.0) >> sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) >> internal_ssl_enable=false >> console_loglevel=debug >> uk-ring=%(400,200,400,450);%(400,2200,400,450) >> us-ring=%(2000, 4000, 440.0, 480.0) >> sip_tls_version=tlsv1 >> fr-ring=%(1500, 3500, 440.0, 0.0) >> bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Tue Jun 16 22:06:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 00:06:30 -0500 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A387874.5090105@gmail.com> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> <4A387874.5090105@gmail.com> Message-ID: can you update and try that again? /b On Jun 17, 2009, at 12:00 AM, paul.degt at gmail.com wrote: > 13564 From darklion11 at yahoo.com Tue Jun 16 22:24:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 22:24:31 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24066849.post@talk.nabble.com> Message-ID: <24067204.post@talk.nabble.com> Yes its already set to false... What should I do? Brian West-3 wrote: > > Its not an error its a warning and you don't have your ACL's > configured correctly. You're trying too hard! :) set auth- > calls=false on the profile. > > /b > > On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote: > >> >> Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on >> FS B > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24067204.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 23:11:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 23:11:31 -0700 (PDT) Subject: [Freeswitch-users] Allow invites from another sip server? Message-ID: <24067552.post@talk.nabble.com> On my acl.conf.xml I allow the ip 116.50.110.2 Is this correct? Error sip_invite() ... Error occur rejected by acl domains param name="apply-inbound-acl" value="domains"/> param name="apply-register_acl" value="domains"/> -- View this message in context: http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From panselva at gmail.com Tue Jun 16 23:14:13 2009 From: panselva at gmail.com (selva kumar) Date: Wed, 17 Jun 2009 11:44:13 +0530 Subject: [Freeswitch-users] outbound error log Message-ID: <45f609f90906162314q3b450f4aid4d98c81af9a1e8f@mail.gmail.com> Hi Michael, I have pasted the freeswitch logs as requested in ( pastebin.freeswitch.org) Thanks Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/0a7867c8/attachment.html From enno.egbert at web.de Tue Jun 16 23:18:45 2009 From: enno.egbert at web.de (NOx-WHV) Date: Tue, 16 Jun 2009 23:18:45 -0700 (PDT) Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> Message-ID: <24067617.post@talk.nabble.com> Hi, look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" where you can ask for special requirements. NOx Diego Viola wrote: > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Tue Jun 16 23:25:19 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 17 Jun 2009 08:25:19 +0200 Subject: [Freeswitch-users] Allow invites from another sip server? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E1E3@cooper> You need to change the apply-inbound-acl to your list (myip) instead of using the domains list. /Peter ----- Ursprungligt meddelande ----- Fr?n: Edmar Cruz Skickat: den 17 juni 2009 08:20 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Allow invites from another sip server? On my acl.conf.xml I allow the ip 116.50.110.2 Is this correct? Error sip_invite() ... Error occur rejected by acl domains param name="apply-inbound-acl" value="domains"/> param name="apply-register_acl" value="domains"/> -- View this message in context: http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a388b4b32938939812364! From grevenx at me.com Tue Jun 16 23:42:12 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 17 Jun 2009 08:42:12 +0200 Subject: [Freeswitch-users] Allow invites from another sip server? In-Reply-To: <24067552.post@talk.nabble.com> References: <24067552.post@talk.nabble.com> Message-ID: I would recomend you to go read on this page: http://wiki.freeswitch.org/wiki/Acl The short answer: replace "domains" with the name of the list you have actually created, that is "myip"... -- Best regards, Even Andr? On 17. juni. 2009, at 08.11, Edmar Cruz wrote: > > On my acl.conf.xml > > I allow the ip 116.50.110.2 > > Is this correct? > > > > > > Error sip_invite() ... Error occur rejected by acl domains > > param name="apply-inbound-acl" value="domains"/> > param name="apply-register_acl" value="domains"/> > > -- > View this message in context: http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yudha2008 at gmail.com Tue Jun 16 23:44:47 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 17 Jun 2009 12:14:47 +0530 Subject: [Freeswitch-users] javascript session.execute Message-ID: *Hi, I have some questions can any one assist me and answer this query Question 1: Can we able to execute all the api command through JavaScript using session.execute. Question 2: How to kill the session using uuid_kill whether it is possible. **If yes means how we will use uuid_kill in session* * i have tried some thing like this : session.execute("uuid_kill", "session.uuid"); session.execute("uuid_kill session.uuid"); but there is no output for the session.execute. Whether it is possible to execute uuid_kill in javascript Thanks in advance Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/4ba25ab3/attachment.html From mcampbellsmith at gmail.com Tue Jun 16 23:58:23 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 17 Jun 2009 16:58:23 +1000 Subject: [Freeswitch-users] Porta Billing? Message-ID: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Hi! Does freeswitch support extracting the billing data (PortaBilling) in SIP messages? If so, is there anyway I can get that information to an extension? 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis? playMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4? bK1FF90 From: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=47E011-580 To: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=adfde4bc91cd85e752cb0672816ac1? a6-eb1b Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB? 7 CSeq: 3 REGISTER PortaBilling: available-funds:7.37 currency:AUD Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060};? expires=3595 Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 Thanks! From jason at jasonjgw.net Wed Jun 17 00:00:19 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 17:00:19 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue Message-ID: <20090617070019.GA5042@jdc.jasonjgw.net> this is an issue which I've been discussing with Brian West on IRC and in e-mail correspondence, which I thought I should bring to the list so that others can look at it as well. The configuration My external SIP profile has its ext-sip-ip and ext-rtp-ip set to stun:stun.freeswitch.org. This is necessary for nat traversal. I have an internal-ipv6 profile as well, which is working, but for some reason it's interfering with calls on the external profile (which of course is an IPv4 profile). The symptom is the following line in outgoing SIP messages while attempting to establish a call to a gateway via the external profile: o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org in response to which, the other side returns a 488 (invalid session description). I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the internal-ipv6.xml profile, but they are said as explained above in the external profile, where the problem lies. This looks like a bug to me but I want to rule out misconfiguration first. This has been tested on revision 13806. From panselva at gmail.com Wed Jun 17 00:02:26 2009 From: panselva at gmail.com (selva kumar) Date: Wed, 17 Jun 2009 12:32:26 +0530 Subject: [Freeswitch-users] Outboubd is not working while inbound is configured and runs well Message-ID: <45f609f90906170002k3938e307ha7f7ab3dd90418de@mail.gmail.com> Hi, I configured oubound in FS, it worked fine. Then I configured inbound in FS,it also worked fine.But now the inbound works fine and the outbound is not working. What is the reason? I got this error logs 2009-06-17 12:11:59 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/1005 at 172.20.192.153[d08bb613-5f40-40c1-8e68-f370177e34a0] 2009-06-17 12:11:59 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing aspro->876159840881185 in context public 2009-06-17 12:11:59 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/internal/1005 at 172.20.192.153[CS_EXECUTE] [NORMAL_CLEARING] 2009-06-17 12:11:59 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 60 (sofia/internal/1005 at 172.20.192.153) Ended 2009-06-17 12:11:59 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/ 1005 at 172.20.192.153 [CS_HANGUP] 2009-06-17 12:16:51 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/9374530072 at 172.20.191.228[d351bf10-2cb9-4cbc-a0fd-66843dc6342b] Can somebody help me on this. -- with regrds Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/196fbc9a/attachment.html From jason at jasonjgw.net Wed Jun 17 00:15:19 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 17:15:19 +1000 Subject: [Freeswitch-users] Outboubd is not working while inbound is configured and runs well In-Reply-To: <45f609f90906170002k3938e307ha7f7ab3dd90418de@mail.gmail.com> References: <45f609f90906170002k3938e307ha7f7ab3dd90418de@mail.gmail.com> Message-ID: <20090617071519.GA9441@jdc.jasonjgw.net> selva kumar wrote: > Hi, > I configured oubound in FS, it worked fine. > Then I configured inbound in FS,it also worked fine.But now the inbound > works fine and the outbound is not working. > What is the reason? If you turn on debug-level logging, it might be possible to work out what is going on. Either run /log debug from fs_cli or make sure that the log level set in your configuration is debug, then capture the logs from your log files or fs_cli and examine the results to find out what is going on. From yivzhenko at mksat.net Wed Jun 17 00:24:15 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Wed, 17 Jun 2009 10:24:15 +0300 Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed In-Reply-To: References: <200906091126.03556.yivzhenko@mksat.net> Message-ID: <200906171024.16279.yivzhenko@mksat.net> Ok, i understand! This variable sets for event CHANNEL_HANGUP_COMPLETE, not for CHANNEL_HANGUP :-) On Tuesday 16 June 2009 19:59:28 you wrote: > That should not be the case - I will double check this. My apologies if I > broke it. :-( > > Please file a bug on this so I don't forget. > > _____ > > From: Yuriy Ivzhenko [mailto:yivzhenko at mksat.net] > Sent: Tuesday, June 09, 2009 1:26 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] mod_nibblebill not set > variablenibble_total_billed > > > > Some time ago mod_nibblebill was set variable nibble_total_billed after > hangup. > > But after last few updates of module this variable is no more sets. > > Somebody else have this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/61c0b326/attachment-0001.html From grevenx at me.com Wed Jun 17 00:26:36 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 17 Jun 2009 09:26:36 +0200 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> References: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Message-ID: <513BEDE8-72A4-4B81-B1F3-6343F13DBB9B@me.com> I'm not sure if you are able to read that header from FS without any modifications. What you can try is the generic syntax to get header variables: {$sip_h_PortaBilling}. Not sure, but I think this might only work with a set of standard headers, as well as all non-standard headers, that really should be prefixed with "X-". If this does not work, the PortaBilling header should in this case be renamed to X-PortaBilling, and you would be able to get it from FreeSWITCH with: {$sip_h_X-PortaBilling}. That is if it's possible to change that header on the server that sends it... Best regards, Even Andr? On 17. juni. 2009, at 08.58, Mark Campbell-Smith wrote: > Hi! > > Does freeswitch support extracting the billing data (PortaBilling) in > SIP messages? If so, is there anyway I can get that information to an > extension? > > 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis playMsg: > Received: > SIP/2.0 200 OK > Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4 bK1FF90 > From: {sip:61XXXXXXXXX at sip.pennytel.com}; tag=47E011-580 > To: {sip:61XXXXXXXXX at sip.pennytel.com}; > tag=adfde4bc91cd85e752cb0672816ac1 a6-eb1b > Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB 7 > CSeq: 3 REGISTER > PortaBilling: available-funds:7.37 currency:AUD > Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060}; expires=3595 > Server: Sip EXpress router (0.9.6 (i386/freebsd)) > Content-Length: 0 > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at suspicious.org Wed Jun 17 00:32:03 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 17 Jun 2009 02:32:03 -0500 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Message-ID: I doubt that header is exposed since its not a standard sip header. However you could probably patch mod_sofia to expose it without too much trouble... How difficult that would be is dependant on where in session that comes in > From: Mark Campbell-Smith > Reply-To: > Date: Wed, 17 Jun 2009 16:58:23 +1000 > To: > Subject: [Freeswitch-users] Porta Billing? > > Hi! > > Does freeswitch support extracting the billing data (PortaBilling) in > SIP messages? If so, is there anyway I can get that information to an > extension? > > 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis? playMsg: > Received: > SIP/2.0 200 OK > Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4? bK1FF90 > From: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=47E011-580 > To: {sip:61XXXXXXXXX at sip.pennytel.com};? > tag=adfde4bc91cd85e752cb0672816ac1? a6-eb1b > Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB? 7 > CSeq: 3 REGISTER > PortaBilling: available-funds:7.37 currency:AUD > Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060};? expires=3595 > Server: Sip EXpress router (0.9.6 (i386/freebsd)) > Content-Length: 0 > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Wed Jun 17 00:36:06 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 17:36:06 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617070019.GA5042@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> Message-ID: <20090617073606.GA13623@jdc.jasonjgw.net> Jason White wrote: > The symptom is the following line in outgoing SIP messages while attempting to > establish a call to a gateway via the external profile: > > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org However, if I place an IPv6 call via the internal-ipv6 profile, the o= line contains the correct IPv6 address - so this is only adversely affecting the IPv4 external profile, it seems. From darklion11 at yahoo.com Wed Jun 17 00:44:54 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 17 Jun 2009 00:44:54 -0700 (PDT) Subject: [Freeswitch-users] Allow invites from another sip server? In-Reply-To: <24067552.post@talk.nabble.com> References: <24067552.post@talk.nabble.com> Message-ID: <1245224694870-3091145.post@n2.nabble.com> Thanks a lot... A mistake from me... Edmar Cruz wrote: > > > On my acl.conf.xml > > I allow the ip 116.50.110.2 > > Is this correct? > > > > > > Error sip_invite() ... Error occur rejected by acl domains > > param name="apply-inbound-acl" value="domains"/> > param name="apply-register_acl" value="domains"/> > > -- > View this message in context: > http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Allow-invites-from-another-sip-server--tp3090843p3091145.html Sent from the freeswitch-users mailing list archive at Nabble.com. From demuel at thephinix.org Wed Jun 17 01:16:57 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Wed, 17 Jun 2009 09:16:57 +0100 (BST) Subject: [Freeswitch-users] Porta Billing? In-Reply-To: References: Message-ID: <036b0e54c43ce10d1d64bb4db2c2eac9.squirrel@www.thephinix.org> Porta-Billing from PortaSIP? I've tried their stuff and this is when I saw the most disgusting piece of scam software on the VoIP world. Their technical support were the dumbest I ever encountered since time immemorial....Be very beware, they are just extracting your money but their services are shit. > I doubt that header is exposed since its not a standard sip header. > However > you could probably patch mod_sofia to expose it without too much > trouble... > How difficult that would be is dependant on where in session that comes in > > >> From: Mark Campbell-Smith >> Reply-To: >> Date: Wed, 17 Jun 2009 16:58:23 +1000 >> To: >> Subject: [Freeswitch-users] Porta Billing? >> >> Hi! >> >> Does freeswitch support extracting the billing data (PortaBilling) in >> SIP messages? If so, is there anyway I can get that information to an >> extension? >> >> 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis? playMsg: >> Received: >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4? bK1FF90 >> From: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=47E011-580 >> To: {sip:61XXXXXXXXX at sip.pennytel.com};? >> tag=adfde4bc91cd85e752cb0672816ac1? a6-eb1b >> Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB? 7 >> CSeq: 3 REGISTER >> PortaBilling: available-funds:7.37 currency:AUD >> Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060};? expires=3595 >> Server: Sip EXpress router (0.9.6 (i386/freebsd)) >> Content-Length: 0 >> >> Thanks! >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Wed Jun 17 01:27:16 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 18:27:16 +1000 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: References: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Message-ID: <20090617082716.GA25006@jdc.jasonjgw.net> Ken Rice wrote: > I doubt that header is exposed since its not a standard sip header. However > you could probably patch mod_sofia to expose it without too much trouble... > How difficult that would be is dependant on where in session that comes in Using the info application will reveal whether that header is available in a channel variable. The event mechanism also discloses the channel variables; you can subscribe to events using fs_cli even without writing a script. From jingwei.yang at gmail.com Wed Jun 17 02:39:33 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Wed, 17 Jun 2009 17:39:33 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> Message-ID: <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> Hi Giovanni, Sorry, pretty busy and fully occupied by other stuff today. Have to delay the testing and give you the result tomorrow. Regards, -Jingwei On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang wrote: > Sure, I'll append to you the result tomorrow. > > Regards, > -Jingwei > > > On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli wrote: > >> Hi Jingwei, >> >> Thanks a lot! I'll take care of as soon as possible. >> >> Btw, before I read the Jira, are you testing in linux? >> >> If you are testing on linux, would you please report how it is >> performing under load? I mean, what is the load average with, let say, >> 10 or 20 or more concurrent Skype call? >> >> This has nothing to do with your bug, but will help me in getting >> better performances. >> >> Ciao for now, and thanks again for reporting! >> >> -giovanni >> >> >> >> >> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >> wrote: >> > Hi Giovanni, >> > >> > I've reported it in Jira. Here's the bug url: >> > >> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >> > >> > Thanks, >> > -Jingwei >> > >> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < >> gmaruzz at celliax.org> >> > wrote: >> >> >> >> Hi Jingwel, >> >> thanks for reporting. >> >> >> >> Could you please add a Jira issue with as much details as possible? >> >> >> >> general guide for reporting bugs: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> >> >> what to add for skypiax: >> >> >> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >> >> >> >> mod_skypiax Jira: >> >> http://jira.freeswitch.org/browse/MODSKYPIAX >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> >> >> >> >> >> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang >> >> wrote: >> >> > Hi Team, >> >> > >> >> > I've been using the record_session feature to record call sessions. >> >> > Here's >> >> > how I prepared the dialplan: >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > And here's how I trigger it: >> >> > >> >> > freeswitch at localhost.localdomain>originate >> skypiax/skypiax2/userAAA >> >> > 2909/userBBB >> >> > >> >> > The call can be established and the data.wav file was generated >> without >> >> > any >> >> > problem. However, once userAAA hung up, a segmentation fault occurred >> >> > and >> >> > freeswitch was automatically shut down. Here are what I got from the >> >> > console: >> >> > >> >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA >> >> > 2909/userBBB >> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >> >> > switch_channel_set_name() >> >> > New Channel skypiax/skypiax2/userAAA >> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >> >> > remote_party_is_ringing() >> >> > Ring-Ready skypiax/skypiax2/userAAA >> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >> >> > outbound_channel_answered() >> >> > Channel [skypiax/skypiax2/userAAA] has been answered >> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >> >> > switch_ivr_session_transfer() >> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >> >> > >> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >> >> > FreeSWITCH->2909/userBBB >> >> > in context default >> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >> >> > switch_channel_set_name() >> >> > New Channel skypiax/ANY/userBBB >> [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >> >> > remote_party_is_ringing() >> >> > Ring-Ready skypiax/ANY/userBBB! >> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >> >> > outbound_channel_answered() >> >> > Channel [skypiax/ANY/userBBB] has been answered >> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >> >> > [CS_EXECUTE] >> >> > [NORMAL_CLEARING] >> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >> Ended >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >> >> > [CS_DESTROY] >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >> >> > [CS_DESTROY] >> >> > Segmentation fault (core dumped) >> >> > >> >> > Please kindly let me know whether there's anything wrong with the >> >> > dialplan >> >> > or the way how I originated the call. >> >> > >> >> > Thanks! >> >> > -Jingwei >> >> > >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/eb3f6ec4/attachment-0001.html From gmaruzz at celliax.org Wed Jun 17 02:58:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 17 Jun 2009 11:58:01 +0200 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> Message-ID: <7b197bef0906170258v1aefcaesd8593737a2e6bc4b@mail.gmail.com> Thanks Jingwei, have a good night! -giovanni On Wed, Jun 17, 2009 at 11:39 AM, Jingwei Yang wrote: > Hi Giovanni, > > Sorry, pretty busy and fully occupied by other stuff today. Have to delay > the testing and give you the result tomorrow. > > Regards, > -Jingwei > > On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang > wrote: >> >> Sure, I'll append to you the result tomorrow. >> >> Regards, >> -Jingwei >> >> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli >> wrote: >>> >>> Hi Jingwei, >>> >>> Thanks a lot! I'll take care of as soon as possible. >>> >>> Btw, before I read the Jira, are you testing in linux? >>> >>> If you are testing on linux, would you please report how it is >>> performing under load? I mean, what is the load average with, let say, >>> 10 or 20 or more concurrent Skype call? >>> >>> This has nothing to do with your bug, but will help me in getting >>> better performances. >>> >>> Ciao for now, and thanks again for reporting! >>> >>> -giovanni >>> >>> >>> >>> >>> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >>> wrote: >>> > Hi Giovanni, >>> > >>> > I've reported it in Jira. Here's the bug url: >>> > >>> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >>> > >>> > Thanks, >>> > -Jingwei >>> > >>> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli >>> > >>> > wrote: >>> >> >>> >> Hi Jingwel, >>> >> thanks for reporting. >>> >> >>> >> Could you please add a Jira issue with as much details as possible? >>> >> >>> >> general guide for reporting bugs: >>> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >> >>> >> what to add for skypiax: >>> >> >>> >> >>> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >>> >> >>> >> mod_skypiax Jira: >>> >> http://jira.freeswitch.org/browse/MODSKYPIAX >>> >> >>> >> >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> ========================================= >>> >> www.celliax.org >>> >> via Pierlombardo 9, 20135 Milano >>> >> Italy >>> >> gmaruzz at celliax dot org >>> >> Cell : +39-347-2665618 >>> >> Fax : +39-02-87390039 >>> >> >>> >> >>> >> >>> >> >>> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang >>> >> wrote: >>> >> > Hi Team, >>> >> > >>> >> > I've been using the record_session feature to record call sessions. >>> >> > Here's >>> >> > how I prepared the dialplan: >>> >> > >>> >> > ??? >>> >> > ????? >> >> > expression="^2909/(.*)$"> >>> >> > ??????? >>> >> > ??????? >>> >> > ????? >>> >> > ??? >>> >> > >>> >> > And here's how I trigger it: >>> >> > >>> >> > ??? freeswitch at localhost.localdomain>originate >>> >> > skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > >>> >> > The call can be established and the data.wav file was generated >>> >> > without >>> >> > any >>> >> > problem. However, once userAAA hung up, a segmentation fault >>> >> > occurred >>> >> > and >>> >> > freeswitch was automatically shut down. Here are what I got from the >>> >> > console: >>> >> > >>> >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/skypiax2/userAAA >>> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >>> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/skypiax2/userAAA >>> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/skypiax2/userAAA] has been answered >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >>> >> > switch_ivr_session_transfer() >>> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >>> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >>> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >>> >> > >>> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >>> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >>> >> > FreeSWITCH->2909/userBBB >>> >> > in context default >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/ANY/userBBB >>> >> > [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >>> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/ANY/userBBB! >>> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/ANY/userBBB] has been answered >>> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >>> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >>> >> > [CS_EXECUTE] >>> >> > [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >>> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >>> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >>> >> > Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >>> >> > [CS_DESTROY] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >>> >> > [CS_DESTROY] >>> >> > Segmentation fault (core dumped) >>> >> > >>> >> > Please kindly let me know whether there's anything wrong with the >>> >> > dialplan >>> >> > or the way how I originated the call. >>> >> > >>> >> > Thanks! >>> >> > -Jingwei >>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From panselva at gmail.com Wed Jun 17 04:19:03 2009 From: panselva at gmail.com (selva kumar) Date: Wed, 17 Jun 2009 16:49:03 +0530 Subject: [Freeswitch-users] Outboubd is not working while inbound and runs well Message-ID: <45f609f90906170419t74033fcmd878658fd109bdfa@mail.gmail.com> Hi When I add the following line in acl.conf.xml file then inbound works fine. But when I comment these lines only the outbound is working fine.What would be problem? By Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/bb2579f2/attachment.html From dujinfang at gmail.com Wed Jun 17 05:49:05 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 17 Jun 2009 20:49:05 +0800 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066825.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> Message-ID: comment lines in the user directory do the trick: On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote: > > If FS A has an account 8011105 does FS B also nid to register > 8011105? Yes it > working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option > to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jim at evolutiontel.net Wed Jun 17 06:14:01 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 17 Jun 2009 23:14:01 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue Message-ID: IMHO, as FS is a B2BUA the new leg should state ownership in the SDP. Add to this the fact the IPV6 is blindly copied from leg 1 and the IP address was not decoded correctly there does appear to be a defficiency in the code. - original message - Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue From: Jason White Date: 17/06/2009 07:01 this is an issue which I've been discussing with Brian West on IRC and in e-mail correspondence, which I thought I should bring to the list so that others can look at it as well. The configuration My external SIP profile has its ext-sip-ip and ext-rtp-ip set to stun:stun.freeswitch.org. This is necessary for nat traversal. I have an internal-ipv6 profile as well, which is working, but for some reason it's interfering with calls on the external profile (which of course is an IPv4 profile). The symptom is the following line in outgoing SIP messages while attempting to establish a call to a gateway via the external profile: o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org in response to which, the other side returns a 488 (invalid session description). I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the internal-ipv6.xml profile, but they are said as explained above in the external profile, where the problem lies. This looks like a bug to me but I want to rule out misconfiguration first. This has been tested on revision 13806. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Jun 17 06:16:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:16:04 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617073606.GA13623@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> Message-ID: <132F594D-AF1B-4331-865D-DDD9AEBE53A8@freeswitch.org> Right which is what you have to do... I haven't been able to reproduce the issue... which is odd. /b On Jun 17, 2009, at 2:36 AM, Jason White wrote: > Jason White wrote: > >> The symptom is the following line in outgoing SIP messages while >> attempting to >> establish a call to a gateway via the external profile: >> >> o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org > > However, if I place an IPv6 call via the internal-ipv6 profile, the > o= line > contains the correct IPv6 address - so this is only adversely > affecting the > IPv4 external profile, it seems. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Wed Jun 17 06:17:46 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 17 Jun 2009 15:17:46 +0200 Subject: [Freeswitch-users] UniMRCP - current status? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB939F@cooper> I can see that the UniMRCP libs have been added to FS lately, I was just wondering about the current status/stability for this implementation? And will this be ported to Windows as well? Just curious - since you guys add more features all the time :) /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/dcc71a34/attachment-0001.html From anthony.minessale at gmail.com Wed Jun 17 06:18:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:18:40 -0500 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> Message-ID: <191c3a030906170618v214fb0e9y4d5c64b724a97200@mail.gmail.com> You should not report bugs on the mailing list. please report your bug to jira http://jira.freeswitch.org http://wiki.freeswitch.org/wiki/Reporting_Bugs Make sure you attach a backtrace of your issue and file it under skypeiax so giovanni can track it. On Wed, Jun 17, 2009 at 4:39 AM, Jingwei Yang wrote: > Hi Giovanni, > > Sorry, pretty busy and fully occupied by other stuff today. Have to delay > the testing and give you the result tomorrow. > > Regards, > -Jingwei > > > On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang wrote: > >> Sure, I'll append to you the result tomorrow. >> >> Regards, >> -Jingwei >> >> >> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli > > wrote: >> >>> Hi Jingwei, >>> >>> Thanks a lot! I'll take care of as soon as possible. >>> >>> Btw, before I read the Jira, are you testing in linux? >>> >>> If you are testing on linux, would you please report how it is >>> performing under load? I mean, what is the load average with, let say, >>> 10 or 20 or more concurrent Skype call? >>> >>> This has nothing to do with your bug, but will help me in getting >>> better performances. >>> >>> Ciao for now, and thanks again for reporting! >>> >>> -giovanni >>> >>> >>> >>> >>> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >>> wrote: >>> > Hi Giovanni, >>> > >>> > I've reported it in Jira. Here's the bug url: >>> > >>> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >>> > >>> > Thanks, >>> > -Jingwei >>> > >>> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < >>> gmaruzz at celliax.org> >>> > wrote: >>> >> >>> >> Hi Jingwel, >>> >> thanks for reporting. >>> >> >>> >> Could you please add a Jira issue with as much details as possible? >>> >> >>> >> general guide for reporting bugs: >>> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >> >>> >> what to add for skypiax: >>> >> >>> >> >>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >>> >> >>> >> mod_skypiax Jira: >>> >> http://jira.freeswitch.org/browse/MODSKYPIAX >>> >> >>> >> >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> ========================================= >>> >> www.celliax.org >>> >> via Pierlombardo 9, 20135 Milano >>> >> Italy >>> >> gmaruzz at celliax dot org >>> >> Cell : +39-347-2665618 >>> >> Fax : +39-02-87390039 >>> >> >>> >> >>> >> >>> >> >>> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang>> > >>> >> wrote: >>> >> > Hi Team, >>> >> > >>> >> > I've been using the record_session feature to record call sessions. >>> >> > Here's >>> >> > how I prepared the dialplan: >>> >> > >>> >> > >>> >> > >> expression="^2909/(.*)$"> >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > And here's how I trigger it: >>> >> > >>> >> > freeswitch at localhost.localdomain>originate >>> skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > >>> >> > The call can be established and the data.wav file was generated >>> without >>> >> > any >>> >> > problem. However, once userAAA hung up, a segmentation fault >>> occurred >>> >> > and >>> >> > freeswitch was automatically shut down. Here are what I got from the >>> >> > console: >>> >> > >>> >> > freeswitch at localhost.localdomain> originate >>> skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/skypiax2/userAAA >>> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >>> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/skypiax2/userAAA >>> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/skypiax2/userAAA] has been answered >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >>> >> > switch_ivr_session_transfer() >>> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >>> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >>> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >>> >> > >>> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >>> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >>> >> > FreeSWITCH->2909/userBBB >>> >> > in context default >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/ANY/userBBB >>> [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >>> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/ANY/userBBB! >>> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/ANY/userBBB] has been answered >>> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >>> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >>> >> > [CS_EXECUTE] >>> >> > [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >>> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >>> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >>> Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >>> >> > [CS_DESTROY] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >>> >> > [CS_DESTROY] >>> >> > Segmentation fault (core dumped) >>> >> > >>> >> > Please kindly let me know whether there's anything wrong with the >>> >> > dialplan >>> >> > or the way how I originated the call. >>> >> > >>> >> > Thanks! >>> >> > -Jingwei >>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/db5354a2/attachment.html From brian at freeswitch.org Wed Jun 17 06:24:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:24:02 -0500 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: References: Message-ID: While the header looks valid it should be an X-Header then it would show up. /b On Jun 17, 2009, at 2:32 AM, Ken Rice wrote: > I doubt that header is exposed since its not a standard sip header. > However > you could probably patch mod_sofia to expose it without too much > trouble... > How difficult that would be is dependant on where in session that > comes in From brian at freeswitch.org Wed Jun 17 06:25:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:25:09 -0500 Subject: [Freeswitch-users] UniMRCP - current status? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB939F@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB939F@cooper> Message-ID: <9584E028-F7D9-4020-9997-2E8680C083BC@freeswitch.org> No, we haven't done the solution file for the module on windows but the lib has been done on windows. It still need more testing and such but its functional. /b On Jun 17, 2009, at 8:17 AM, Peter Olsson wrote: > I can see that the UniMRCP libs have been added to FS lately, I was > just wondering about the current status/stability for this > implementation? And will this be ported to Windows as well? > > Just curious ? since you guys add more features all the time :) > > /Peter > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/1ca4439c/attachment-0001.html From brian at freeswitch.org Wed Jun 17 06:26:13 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:26:13 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: References: Message-ID: <931BB0A4-DF71-4E3C-9968-8AED09C46DFF@freeswitch.org> On Jun 17, 2009, at 8:14 AM, Jim Burke wrote: > IMHO, as FS is a B2BUA the new leg should state ownership in the > SDP. Add to this the fact the IPV6 is blindly copied from leg 1 and > the IP address was not decoded correctly there does appear to be a > defficiency in the code. I don't think that is what is going on unless you're trying to do proxy media from IPv4 to IPv6 which I haven't ever tried nor do I recommend. /b From anthony.minessale at gmail.com Wed Jun 17 06:27:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:27:03 -0500 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: References: Message-ID: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> 1) look in the wiki and try to understand the difference between application and FSAPI functions. 2) use apiExecute function from js to execute FSAPI functions such as uuid_kill On Wed, Jun 17, 2009 at 1:44 AM, Baskar wrote: > *Hi, > > I have some questions can any one assist me and answer this query > > Question 1: Can we able to execute all the api command through JavaScript > using session.execute. > > Question 2: How to kill the session using uuid_kill whether it is > possible. **If yes means how we will use uuid_kill in session* > * > i have tried some thing like this : > > session.execute("uuid_kill", "session.uuid"); > > session.execute("uuid_kill session.uuid"); > > but there is no output for the session.execute. > > Whether it is possible to execute uuid_kill in javascript > > Thanks in advance > > > Warm Regards, > N.Baskar > > * > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/bf13cb23/attachment.html From anthony.minessale at gmail.com Wed Jun 17 06:32:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:32:42 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617070019.GA5042@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> Message-ID: <191c3a030906170632s39130c29m717324660f0d0fbb@mail.gmail.com> can you confirm that they are not being inherited by some of the helper variables exported from vars.xml? can you minimalise your ip4 profile by cutting all the commented lines and actually remove the ext-rtp-ip and ext-sip-ip params? Then possibly post your config for each profile. On Wed, Jun 17, 2009 at 2:00 AM, Jason White wrote: > this is an issue which I've been discussing with Brian West on IRC and in > e-mail correspondence, which I thought I should bring to the list so that > others can look at it as well. > > The configuration > > My external SIP profile has its ext-sip-ip and ext-rtp-ip set to > stun:stun.freeswitch.org. This is necessary for nat traversal. > > I have an internal-ipv6 profile as well, which is working, but for some > reason > it's interfering with calls on the external profile (which of course is an > IPv4 profile). > > The symptom is the following line in outgoing SIP messages while attempting > to > establish a call to a gateway via the external profile: > > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org > > in response to which, the other side returns a 488 (invalid session > description). > > I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the > internal-ipv6.xml profile, but they are said as explained above in the > external profile, where the problem lies. > > This looks like a bug to me but I want to rule out misconfiguration first. > > This has been tested on revision 13806. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/abb0e0c2/attachment.html From brian at freeswitch.org Wed Jun 17 06:36:22 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:36:22 -0500 Subject: [Freeswitch-users] outbound error log In-Reply-To: <45f609f90906162314q3b450f4aid4d98c81af9a1e8f@mail.gmail.com> References: <45f609f90906162314q3b450f4aid4d98c81af9a1e8f@mail.gmail.com> Message-ID: The link would be helpful. /b On Jun 17, 2009, at 1:14 AM, selva kumar wrote: > > Hi Michael, > I have pasted the freeswitch logs as requested in > (pastebin.freeswitch.org) > > > Thanks > Sam > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/aee53dee/attachment.html From anthony.minessale at gmail.com Wed Jun 17 06:36:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:36:35 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37FC59.60401@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> <4A37DB1C.8070706@gmx.net> <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> <4A37FC59.60401@gmx.net> Message-ID: <191c3a030906170636s275378bey40ce3eca6b5d99a4@mail.gmail.com> Looks like you are trying to build a call center have you seen mod_fifo? It's designed to let people on headsets sit idle and you can send calls to them at will. On Tue, Jun 16, 2009 at 3:11 PM, Peter P GMX wrote: > Thanks Michael, > > I have disabled it now. > > I finally got it to work, (sip_h_Call-Info=;answer-after=0) > but the behaviour was not as desired, as I didn't manage the phone to > pick up the call on the headset. It will only have the speaker enabled. > So I will have to go a different way with parking the call and then > forward it. > > Best regards > Peter > > > Michael Jerris schrieb: > > uuid_setvar sip_invite_params intercom=true should be > > unnecessary. > > > > Mike > > > > On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: > > > > > >> It mainly works now by uuid_transfer the following way via event > >> socket. > >> uuid_setvar sip_invite_params intercom=true > >> uuid_setvar sip_auto_answer true > >> uuid_transfer 1000 XML default > >> so the call is transferred from 1000 to 1000. > >> > >> What happens: > >> 1) If I disable intercom on the Snom phone, the phone rings, stops > >> ringing and rings again (ok) > >> 1) If I enable intercom on the Snom phone, the phone rings, stops > >> ringing and hangs up (not ok) > >> > >> So I do not get the Snom to pick up the call in intercom mode. > >> > >> The last invite is: > >> INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib > >> SIP/2.0 > >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > >> Route: ;transport=tls;line=er6kxnib > >> Max-Forwards: 68 > >> From: "Peter FS" ;tag=9eQ8rjQy533HF > >> To: > >> >> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > >> CSeq: 116467629 INVITE > >> Contact: > >> Call-Info: ;answer-after=0 > >> The intercom part is there and the Call-Info line with answer-after > >> also. > >> > >> The phone answers with > >> SIP/2.0 401 Unauthorized > >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > >> From: "Peter FS" ;tag=9eQ8rjQy533HF > >> To: > >> >> > >>> ;tag=71rskygkr2 > >>> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > >> CSeq: 116467629 INVITE > >> Contact: > >> ;reg-id=1 > >> WWW-Authenticate: Digest realm="sip2.mycompany.de", > >> nonce="2ee26efe6ab27f88", algorithm=MD5 > >> Content-Length: 0 > >> and hangs up. > >> > >> Anybody know how to solve this Snom intercom issue? > >> > >> Best regards > >> Peter > >> > >> > >> Michael Jerris schrieb: > >> > >>> The transfer should work but it sounds like offhook agents is what > >>> your really trying to accomplish so I would go with brian's > >>> suggestion. > >>> > >>> > >>> > >>> On Jun 16, 2009, at 7:38 AM, Peter P GMX > >>> wrote: > >>> > >>> > >>> > >>>> Hello Michael, > >>>> > >>>> I want the phone be ringing, just for acoustical feedback reasons. > >>>> > >>>> But what if I > >>>> > >>>> * transfer it to the same user destination again (now with > >>>> intercom > >>>> enabled), will this work? > >>>> * transfer it to park and then transfer it to the same destination > >>>> again (now with intercom enabled) > >>>> > >>>> Best regards > >>>> Peter > >>>> > >>>> Michael Jerris schrieb: > >>>> > >>>> > >>>>> The only way I can think to do this today would be to cancel the > >>>>> call > >>>>> and re send with the intercom headers for a phone that supports it. > >>>>> It may be possible to send a reinvite with autoanswer headers but I > >>>>> doubt that would work, all you could do is try making code to do it > >>>>> it > >>>>> a sipp or sipsak scenario and test it. A better aproach might be > >>>>> to > >>>>> answer the call normally and detect that to start your web workflow > >>>>> or > >>>>> not really ring the phone, just the web app and deliver the call > >>>>> with > >>>>> autoanswer when the button is hit in the web ui. > >>>>> > >>>>> Mike > >>>>> > >>>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX > >>>>> wrote: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>> Hello Brian, > >>>>>> > >>>>>> this is too easy :-). > >>>>>> > >>>>>> This is for a small callcenter app and I only want the user to > >>>>>> pickup > >>>>>> the call once (to accept the call in X-Lite (or a Snom phone) > >>>>>> and to > >>>>>> start the workflow on the web application). I do not want him to > >>>>>> accept > >>>>>> the call on the phone and then on the Web app. > >>>>>> > >>>>>> Best regards > >>>>>> Peter > >>>>>> > >>>>>> > >>>>>> > >>>>>> Brian West schrieb: > >>>>>> > >>>>>> > >>>>>> > >>>>>>> click on the AA button? :) > >>>>>>> > >>>>>>> /b > >>>>>>> > >>>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>> What is the best way to have this done? Move the call to park > >>>>>>>> and > >>>>>>>> then > >>>>>>>> retransfer again with intercom, or is there a better solution? > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>> _______________________________________________ > >>>>>>> Freeswitch-users mailing list > >>>>>>> Freeswitch-users at lists.freeswitch.org > >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>>> http://www.freeswitch.org > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> _______________________________________________ > >>>>>> Freeswitch-users mailing list > >>>>>> Freeswitch-users at lists.freeswitch.org > >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> http://www.freeswitch.org > >>>>>> > >>>>>> > >>>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/6f9ffc12/attachment-0001.html From rdenert at tng.de Wed Jun 17 06:42:41 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 17 Jun 2009 15:42:41 +0200 (CEST) Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <15918673.24821245245881356.JavaMail.root@zimbra.tng.de> Message-ID: <12807519.24911245246161312.JavaMail.root@zimbra.tng.de> Hello, I have one question about including a webserver. I made a dialplan which I deposited on a webserver. That was no the problem. But I have a commandline in the dialplan that doesn't work correct. The fs_cli says: (this line has no problem. I just mentioned this for the sake of completeness) My features.xml looks like this: In my freeswitch server this line works perfekt. The extension "test" in features.xml is executed. For near information: When I call the FS (it is a conferencing system) the machine looks in the dialplan which is deposited on a webserver. Than I press the digits into phone to get access in the conferencing room. But than I get two important messages in the fs_cli: 2009-06-17 15:38:21 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 123456789->test in context features 2009-06-17 15:38:21 [WARNING] mod_dialplan_xml.c:263 dialplan_hunt() Context features not found Does anybody has an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From brian at freeswitch.org Wed Jun 17 06:47:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:47:57 -0500 Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <12807519.24911245246161312.JavaMail.root@zimbra.tng.de> References: <12807519.24911245246161312.JavaMail.root@zimbra.tng.de> Message-ID: <78F65C7F-F4E6-4CB4-ADEE-3557385E8039@freeswitch.org> Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > Context features not found From timb0311 at hotmail.com Wed Jun 17 07:00:25 2009 From: timb0311 at hotmail.com (Tim B) Date: Wed, 17 Jun 2009 10:00:25 -0400 Subject: [Freeswitch-users] Mod_Fax / TxFax / Originate In-Reply-To: References: Message-ID: Trying to do a local test for faxing. Keep getting an error. default dialplan: //inbound from remote box works fine - connect asterisk box and fs box, then fax from asterisk to fs... OK - also fax from fs to asterisk.... OK // local fax on fs .... FAILS!! // my originate command: originate sofia/internal/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) // error as follows: 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing FreeSWITCH->8000 in context public 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 Legged calls 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] _________________________________________________________________ Lauren found her dream laptop. Find the PC that?s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/81096a52/attachment.html From krice at suspicious.org Wed Jun 17 08:15:44 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 17 Jun 2009 10:15:44 -0500 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: <20090617082716.GA25006@jdc.jasonjgw.net> Message-ID: While you are correct Jason, this particular header is not a valid SIP Header... It should be prefixed with X- ie: "X-Some-Custom-Header: foo" When you have non-standard headers lib sofia will stick them in a struct on the backend for us to manually deal with but they are not auto-magically assigned to channel variables unless they are X- and P- headers (P- headers are some special meaning headers typically associated with billing and privacy or are "private" headers > From: Jason White > > Using the info application will reveal whether that header is available in a > channel variable. > > The event mechanism also discloses the channel variables; you can subscribe to > events using fs_cli even without writing a script. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonny.voip at gmail.com Wed Jun 17 08:19:35 2009 From: jonny.voip at gmail.com (Jonathan DiVita) Date: Wed, 17 Jun 2009 11:19:35 -0400 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita) In-Reply-To: References: Message-ID: I was able to compile mod_opal with the patches you suggested. Now, once that is done is there anything more I need to do inorder for mod_opal to show up when I run freeswitch? I didn't see that it had started when I ran freeswitch after the mod_opal compile. Thanks! Jon ----- Original Message ----- From: To: Sent: Wednesday, June 17, 2009 12:26 AM Subject: Freeswitch-users Digest, Vol 36, Issue 159 > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Compiling Issues: Opal with Latest SVN Builds 6-19-09 > (Brian West) > 2. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 3. Re: session.getDigits() not working (Brian West) > 4. Re: How can I join two freeswitch on two servers? (Brian West) > 5. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 6. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 7. Re: How can I join two freeswitch on two servers? (Brian West) > 8. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 16 Jun 2009 22:12:01 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN > Builds 6-19-09 > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="iso-8859-1" > > please see MODOPAL-10 on jira. > > /b > > On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > >> Hello, all. I'm currently playing around with a new install of >> Freeswitch and wanted to try out mod_opal. Below are the current >> SVN builds for opal, ptlib, and freeswitch. I end up with the >> following errors when compiling. >> >> making all mod_opal >> Compiling mod_opal.cpp... >> Compiling mod_opal.cpp ... >> In file included from mod_opal.cpp:25: >> mod_opal.h:151: error: conflicting return type specified for >> ?virtual OpalLocalConnection* >> FSEndPoint::CreateConnection(OpalCall&, void*)? >> /usr/include/opal/opal/localep.h:267: error: overriding ?virtual >> ptlib_virtual_function_changed_or_removed****** >> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? >> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, >> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, >> switch_channel_t*)?: >> mod_opal.cpp:564: error: no matching function for call to >> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, >> NULL)? >> /usr/include/opal/opal/localep.h:290: note: candidates are: >> OpalLocalConnection::OpalLocalConnection(OpalCall&, >> OpalLocalEndPoint&, void*, unsigned int, >> OpalConnection::StringOptions*, char) >> /usr/include/opal/opal/localep.h:276: note: >> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) >> make[4]: *** [mod_opal.lo] Error 1 >> make[3]: *** [all] Error 1 >> make[2]: *** [mod_opal-all] Error 1 >> make[1]: *** [mod_opal] Error 2 >> make: *** [mod_opal] Error 2 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <929908.28142.qm at web57310.mail.re1.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > How can i turn off authentication? This is my acl.conf.xml on > 192.168.0.105 > > > > > > > On 192.168.0.4 > > > > > > > ________________________________ > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, June 16, 2009 10:49:58 PM > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > > Turn off authentication or use ACL's > > /b > > On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > >> Is there another way to manage the gateway with the caller id of the >> user >> not the gateway user id and is there a gateway that doesn't need a >> username >> and password? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/2839a86d/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Tue, 16 Jun 2009 22:31:21 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] session.getDigits() not working > To: freeswitch-users at lists.freeswitch.org > Message-ID: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Shouldn't have really changed any behavior at all... What svn rev are > you on? > > /b > > On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > >> API CALL [global_getvar()] output: >> external_ssl_enable=false >> external_tls_port=5081 >> external_sip_port=5080 >> external_auth_calls=false >> internal_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_sip_port=5060 >> default_provider_contact=5000 >> default_provider_from_domain=example.com >> default_provider_password=password >> external_rtp_ip=74.92.196.241 >> xmpp_server_profile=xmpps >> xmpp_client_profile=xmppc >> global_codec_prefs=G722,PCMU,PCMA,GSM >> hold_music=local_stream://moh >> external_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_auth_calls=true >> local_ip_v4=192.168.0.40 >> unroll_loops=true >> default_areacode=918 >> default_provider_register=false >> local_mask_v4=255.255.255.0 >> default_password=1234 >> call_debug=false >> local_ip_v6=::1 >> default_provider_username=joeuser >> sound_prefix=/var/opt/freeswitch/sounds/en/us/callie >> outbound_caller_id=0000000000 >> default_country=US >> base_dir=/var/opt/freeswitch >> bind_server_ip=auto >> internal_tls_port=5061 >> switch_serial=c0a8002854db >> default_provider=example.com >> outbound_codec_prefs=PCMU,PCMA,GSM >> domain_name=192.168.0.40 >> domain=192.168.0.40 >> external_sip_ip=74.92.196.241 >> outbound_caller_name=Versafon.com >> rs-ring=%(1000, 4000, 425.0, 0.0) >> sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) >> internal_ssl_enable=false >> console_loglevel=debug >> uk-ring=%(400,200,400,450);%(400,2200,400,450) >> us-ring=%(2000, 4000, 440.0, 480.0) >> sip_tls_version=tlsv1 >> fr-ring=%(1500, 3500, 440.0, 0.0) >> bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) >> > > > > > ------------------------------ > > Message: 4 > Date: Tue, 16 Jun 2009 22:36:38 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/47e21916/attachment-0001.html > > ------------------------------ > > Message: 5 > Date: Tue, 16 Jun 2009 21:00:04 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 6 > Date: Tue, 16 Jun 2009 21:02:21 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > Like this? > > I put this on external profile > > "/> > "/> > > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 7 > Date: Tue, 16 Jun 2009 23:08:46 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <93456DA1-47C0-4524-903B-0FDE310EE93D at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > COPY paste fail :) > > > > something like that as per the example. > > /b > > On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > >> >> How can sofia profile can call ACL? >> Can you give me an example? >> Like this? >> >> I put this on external profile >> >> "/> >> "/> >> >> >> Brian West-3 wrote: >>> >>> Now you have to tell the sofia profile to use that ACL >>> >>> /b >>> >>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>> >>>> How can i turn off authentication? This is my acl.conf.xml on >>>> 192.168.0.105 >>>> >>>> >>>> >>>> >>>> >>>> On 192.168.0.4 >>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > ------------------------------ > > Message: 8 > Date: Tue, 16 Jun 2009 21:26:38 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066825.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > If FS A has an account 8011105 does FS B also nid to register 8011105? Yes > it > working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option > to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 159 > ************************************************* From Claudio.Cavalera at italtel.it Wed Jun 17 08:20:19 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 17 Jun 2009 17:20:19 +0200 Subject: [Freeswitch-users] Freeswitch as a B2B Application Server for IMS Message-ID: Hello freeswitchers, lately I'm trying to set up a testbed to ivestigate a potential use of freeswitch as a Back-to-Back application server in an IMS architecture. I've seen IMS specs are also linked here http://wiki.freeswitch.org/wiki/Documentation so I've thinked maybe there's a chance :-) I also have the inner feeling that fs could do an amazing job in IMS field as he does in NGN. For my testing I'm now using OpenIMSCore as control layer (where phones do register) and I'm trying to put fs on top of it as a b2b application server to provide services. I would like to share with you my experience to see if something could be done about this scenario (or if it's worth the trouble at least). Alice and Bob are two users registered to the IMS Core and they both have a profile for which originating and terminating INVITEs get triggered towards the fs application server. When Alice calls Bob the call setup would include three legs: 1) Alice -> PCSCF (orig) -> SCSCF (orig) -> FS (orig) 2) FS (orig) -> SCSCF (orig) -> ICSCF -> SCSCF (term) -> FS (term) 3) FS (term) -> SCSCF (term) -> PCSCF(term) -> Bob This partially works already with fs out of the box, but there are a still a few issues to be solved. When FS starts the brand new leg 2) as a B2B User Agent he should keep the Route: header in the SIP INVITE "almost the same" as the one he received in the leg 1) I see here two different issues: a) Getting the Route: header out of incoming invite in leg 1) b) Writing the proper Route: header and have FS behaving correctly at transport layer in the outgoing INVITE in leg 2) a) Now please correct me if I'm wrong: at the moment the header is not a channel variable available in fs (e.g. I don't get it with the "info" app). It there were a way to get this header out of the incoming INVITEs, I could do the logic to parse it and forge a proper one in the outgoing INVITE. b) Concerning how to write the header, I'm already working with fs_path directive which also makes FS behaves correctly at network layer. Could someone please elaborate a little bit about the alternative to fs_path directive? I've seen there are already many in theory: - combining sip_h_Route= with http://wiki.freeswitch.org/wiki/Variable_sip_network_destination - use of http://wiki.freeswitch.org/wiki/Variable_sip_route_uri - use of fs_path= http://wiki.freeswitch.org/wiki/Sofia#Specifying_SIP_Proxy_With_fs_path I've simplified the scenario a little bit, there are other things that the B2B AS should do (e.g. removing Record-Route:) but FS do them already from what I've tested. If anyone in the community is interested I'm here to provide further information or share my experience to implement the best solution. Best regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From peter.olsson at visionutveckling.se Wed Jun 17 08:29:44 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 17 Jun 2009 17:29:44 +0200 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita) In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB93DF@cooper> If you use latest stable opal package you will be able to compile without patching mod_opal. That is what the guys from Opal recommend. The latest trunk has lots of things going on, so I'm not sure the patches I have supplied are sufficient anymore.. To load the module you must also enable it as a module in modules.conf.xml. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jonathan DiVita Skickat: den 17 juni 2009 17:20 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita) I was able to compile mod_opal with the patches you suggested. Now, once that is done is there anything more I need to do inorder for mod_opal to show up when I run freeswitch? I didn't see that it had started when I ran freeswitch after the mod_opal compile. Thanks! Jon ----- Original Message ----- From: To: Sent: Wednesday, June 17, 2009 12:26 AM Subject: Freeswitch-users Digest, Vol 36, Issue 159 > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Compiling Issues: Opal with Latest SVN Builds 6-19-09 > (Brian West) > 2. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 3. Re: session.getDigits() not working (Brian West) > 4. Re: How can I join two freeswitch on two servers? (Brian West) > 5. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 6. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 7. Re: How can I join two freeswitch on two servers? (Brian West) > 8. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 16 Jun 2009 22:12:01 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN > Builds 6-19-09 > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="iso-8859-1" > > please see MODOPAL-10 on jira. > > /b > > On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > >> Hello, all. I'm currently playing around with a new install of >> Freeswitch and wanted to try out mod_opal. Below are the current >> SVN builds for opal, ptlib, and freeswitch. I end up with the >> following errors when compiling. >> >> making all mod_opal >> Compiling mod_opal.cpp... >> Compiling mod_opal.cpp ... >> In file included from mod_opal.cpp:25: >> mod_opal.h:151: error: conflicting return type specified for >> ?virtual OpalLocalConnection* >> FSEndPoint::CreateConnection(OpalCall&, void*)? >> /usr/include/opal/opal/localep.h:267: error: overriding ?virtual >> ptlib_virtual_function_changed_or_removed****** >> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? >> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, >> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, >> switch_channel_t*)?: >> mod_opal.cpp:564: error: no matching function for call to >> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, >> NULL)? >> /usr/include/opal/opal/localep.h:290: note: candidates are: >> OpalLocalConnection::OpalLocalConnection(OpalCall&, >> OpalLocalEndPoint&, void*, unsigned int, >> OpalConnection::StringOptions*, char) >> /usr/include/opal/opal/localep.h:276: note: >> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) >> make[4]: *** [mod_opal.lo] Error 1 >> make[3]: *** [all] Error 1 >> make[2]: *** [mod_opal-all] Error 1 >> make[1]: *** [mod_opal] Error 2 >> make: *** [mod_opal] Error 2 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <929908.28142.qm at web57310.mail.re1.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > How can i turn off authentication? This is my acl.conf.xml on > 192.168.0.105 > > > > > > > On 192.168.0.4 > > > > > > > ________________________________ > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, June 16, 2009 10:49:58 PM > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > > Turn off authentication or use ACL's > > /b > > On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > >> Is there another way to manage the gateway with the caller id of the >> user >> not the gateway user id and is there a gateway that doesn't need a >> username >> and password? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/2839a86d/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Tue, 16 Jun 2009 22:31:21 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] session.getDigits() not working > To: freeswitch-users at lists.freeswitch.org > Message-ID: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Shouldn't have really changed any behavior at all... What svn rev are > you on? > > /b > > On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > >> API CALL [global_getvar()] output: >> external_ssl_enable=false >> external_tls_port=5081 >> external_sip_port=5080 >> external_auth_calls=false >> internal_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_sip_port=5060 >> default_provider_contact=5000 >> default_provider_from_domain=example.com >> default_provider_password=password >> external_rtp_ip=74.92.196.241 >> xmpp_server_profile=xmpps >> xmpp_client_profile=xmppc >> global_codec_prefs=G722,PCMU,PCMA,GSM >> hold_music=local_stream://moh >> external_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_auth_calls=true >> local_ip_v4=192.168.0.40 >> unroll_loops=true >> default_areacode=918 >> default_provider_register=false >> local_mask_v4=255.255.255.0 >> default_password=1234 >> call_debug=false >> local_ip_v6=::1 >> default_provider_username=joeuser >> sound_prefix=/var/opt/freeswitch/sounds/en/us/callie >> outbound_caller_id=0000000000 >> default_country=US >> base_dir=/var/opt/freeswitch >> bind_server_ip=auto >> internal_tls_port=5061 >> switch_serial=c0a8002854db >> default_provider=example.com >> outbound_codec_prefs=PCMU,PCMA,GSM >> domain_name=192.168.0.40 >> domain=192.168.0.40 >> external_sip_ip=74.92.196.241 >> outbound_caller_name=Versafon.com >> rs-ring=%(1000, 4000, 425.0, 0.0) >> sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) >> internal_ssl_enable=false >> console_loglevel=debug >> uk-ring=%(400,200,400,450);%(400,2200,400,450) >> us-ring=%(2000, 4000, 440.0, 480.0) >> sip_tls_version=tlsv1 >> fr-ring=%(1500, 3500, 440.0, 0.0) >> bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) >> > > > > > ------------------------------ > > Message: 4 > Date: Tue, 16 Jun 2009 22:36:38 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/47e21916/attachment-0001.html > > ------------------------------ > > Message: 5 > Date: Tue, 16 Jun 2009 21:00:04 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 6 > Date: Tue, 16 Jun 2009 21:02:21 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > Like this? > > I put this on external profile > > "/> > "/> > > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 7 > Date: Tue, 16 Jun 2009 23:08:46 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <93456DA1-47C0-4524-903B-0FDE310EE93D at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > COPY paste fail :) > > > > something like that as per the example. > > /b > > On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > >> >> How can sofia profile can call ACL? >> Can you give me an example? >> Like this? >> >> I put this on external profile >> >> "/> >> "/> >> >> >> Brian West-3 wrote: >>> >>> Now you have to tell the sofia profile to use that ACL >>> >>> /b >>> >>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>> >>>> How can i turn off authentication? This is my acl.conf.xml on >>>> 192.168.0.105 >>>> >>>> >>>> >>>> >>>> >>>> On 192.168.0.4 >>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > ------------------------------ > > Message: 8 > Date: Tue, 16 Jun 2009 21:26:38 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066825.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > If FS A has an account 8011105 does FS B also nid to register 8011105? Yes > it > working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option > to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 159 > ************************************************* _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a390b1b32934139097269! From paul.degt at gmail.com Wed Jun 17 08:52:31 2009 From: paul.degt at gmail.com (paul.degt) Date: Wed, 17 Jun 2009 11:52:31 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> <4A387874.5090105@gmail.com> Message-ID: <4A39113F.8080702@gmail.com> I will in a couple of days and report back. Brian West wrote: > can you update and try that again? > > /b > > On Jun 17, 2009, at 12:00 AM, paul.degt at gmail.com wrote: > > >> 13564 >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From larclap at yahoo.com Wed Jun 17 10:06:50 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 17 Jun 2009 10:06:50 -0700 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? Message-ID: <012901c9ef6e$01b3ee60$051bcb20$@com> In conf/dialplan/default.xml, the eavesdrop extension's condition is - expression="^88(.*)$|^\*0(.*)$"> Is this intended? I thought it was defined to eavesdrop on internal extensions. Why wouldn't it be something like - expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop extension. Log: 1030 Dialplan: sofia/internal/1009 at 192.168.10.29 parsing [default->eavesdrop] continue=false 1031 Dialplan: sofia/internal/1009 at 192.168.10.29 Regex (PASS) [eavesdrop] destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false 1032 Dialplan: sofia/internal/1009 at 192.168.10.29 Action answer() Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/947d075d/attachment-0001.html From msc at freeswitch.org Wed Jun 17 10:22:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Jun 2009 12:22:46 -0500 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? In-Reply-To: <012901c9ef6e$01b3ee60$051bcb20$@com> References: <012901c9ef6e$01b3ee60$051bcb20$@com> Message-ID: <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb wrote: > In conf/dialplan/default.xml, the eavesdrop extension's condition is - > expression="^88(.*)$|^\*0(.*)$"> > > > > Is this intended? I thought it was defined to eavesdrop on internal > extensions. Why wouldn't it be something like - > expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial > 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop > extension. > > > Lars, "8885551212" should never match regex expression "^88(\d{3,5})$" . It looks to me like your eavesdrop is matching on "^88(.*)$" which most definitely WILL match 8885551212. Please check your dialplan's eavesdrop extension's regex and make sure it is correct. -MC > Log: > > 1030 Dialplan: sofia/internal/1009 at 192..168.10.29parsing [default->eavesdrop] continue=false > > 1031 Dialplan: sofia/internal/1009 at 192..168.10.29Regex (PASS) [eavesdrop] destination_number(8885819795) =~ > /^88(.*)$|^\*0(.*)$/ break=on-false > > 1032 Dialplan: sofia/internal/1009 at 192..168.10.29Action answer() > > > > Thanks, Lars > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/d982db85/attachment.html From diego.viola at gmail.com Wed Jun 17 10:26:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 17 Jun 2009 13:26:20 -0400 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> References: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> Message-ID: <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> Applications are the ones in mod_dptools and FSAPI are mod_commands API right? On Wed, Jun 17, 2009 at 9:27 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > 1) look in the wiki and try to understand the difference between > application and FSAPI functions. > 2) use apiExecute function from js to execute FSAPI functions such as > uuid_kill > > > > On Wed, Jun 17, 2009 at 1:44 AM, Baskar wrote: > >> *Hi, >> >> I have some questions can any one assist me and answer this query >> >> Question 1: Can we able to execute all the api command through JavaScript >> using session.execute. >> >> Question 2: How to kill the session using uuid_kill whether it is >> possible. **If yes means how we will use uuid_kill in session* >> * >> i have tried some thing like this : >> >> session.execute("uuid_kill", "session.uuid"); >> >> session.execute("uuid_kill session.uuid"); >> >> but there is no output for the session.execute. >> >> Whether it is possible to execute uuid_kill in javascript >> >> Thanks in advance >> >> >> Warm Regards, >> N.Baskar >> >> * >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/64310a4b/attachment.html From brian at freeswitch.org Wed Jun 17 10:29:56 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 12:29:56 -0500 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> References: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> Message-ID: <9D79D650-31E6-4119-8959-A434CE866C87@freeswitch.org> Thats one way to put it ;) /b On Jun 17, 2009, at 12:26 PM, Diego Viola wrote: > Applications are the ones in mod_dptools and FSAPI are mod_commands > API right? From brian at freeswitch.org Wed Jun 17 11:05:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:05:12 -0500 Subject: [Freeswitch-users] Fwd: [UniMRCP] Open source ASR and TTS plugins References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> Message-ID: <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Guys I'm tossing in $250 dollars of my own money on this ... who is going to pitch in? Arsen and I have been talking about how to accomplish this so we'll have an Open Source Speech Server via MRCP. Who wants to pitch in please paypal brian at freeswitch.org and I can wire it to Arsen. Thanks, Brian Begin forwarded message: > ----- Forwarded Message ---- > From: Arsen Chaloyan > To: UniMRCP > Sent: Wednesday, June 17, 2009 10:57:30 PM > Subject: [UniMRCP] Open source ASR and TTS plugins > > Anybody interested in the development of open source ASR and TTS > plugins for UniMRCP server write me offlist. > > PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ > Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ > > Thanks, > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/62f1371d/attachment.html From bjbrashier at gmail.com Wed Jun 17 11:18:34 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Wed, 17 Jun 2009 11:18:34 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> Message-ID: <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> Well, since what I really need at this time is only about 5 commands of similar complexity to a toggle on something already extant, I've decided to just modify source. I can't imagine that people will be terribly interested in my modifications, but I know I'm interested in being able to stay updated with the current trunk, so I'll have to figure out how to deal with that. I'll let you know if I have trouble. On Tue, Jun 16, 2009 at 5:04 PM, William Suffill wrote: > It depends pretty heavily on what you are trying to add function wise. If > it's more in depth using the event socket would allow it to be used on any > FreeSwitch server assuming it caught the dtmf and acted according without > having to modify the core source code/recompile. It might be a bit more work > at first but could be well worth it depending on your needs. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/42c65616/attachment.html From msc at freeswitch.org Wed Jun 17 11:26:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Jun 2009 13:26:09 -0500 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: <9D79D650-31E6-4119-8959-A434CE866C87@freeswitch.org> References: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> <9D79D650-31E6-4119-8959-A434CE866C87@freeswitch.org> Message-ID: <87f2f3b90906171126l13cabd0atf0aa42a282385d56@mail.gmail.com> Another way to view it: FSAPI or "APIs" are done at the CLI. (API at the CLI) dptools or dial plan applications (apps) are done inside the dialplan So if you have something like "api.execute" then you would use that to execute an FSAPI, just as if you'd typed it at the CLI. On the other hand if you have a session object that's like having a dialplan in your script, so you do dialplan type stuff with a session object. Hope that helps... -MC On Wed, Jun 17, 2009 at 12:29 PM, Brian West wrote: > Thats one way to put it ;) > > /b > > On Jun 17, 2009, at 12:26 PM, Diego Viola wrote: > > > Applications are the ones in mod_dptools and FSAPI are mod_commands > > API right? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/9148f268/attachment-0001.html From shaheryarkh at googlemail.com Wed Jun 17 11:43:36 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 18 Jun 2009 00:43:36 +0600 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia Message-ID: Hi, Is it possible to enable compact SIP headers in mod_sofia configuration? If yes, then how to do so? Kindly give an example. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/6747580f/attachment.html From mitul at enterux.com Wed Jun 17 11:50:52 2009 From: mitul at enterux.com (Mitul Limbani) Date: Thu, 18 Jun 2009 00:20:52 +0530 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: Brain, I can chip in $150 of my own as well, but two things: 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work on it?) 2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 17-Jun-09, at 23:35, Brian West wrote: > Guys I'm tossing in $250 dollars of my own money on this ... who is > going to pitch in? Arsen and I have been talking about how to > accomplish this so we'll have an Open Source Speech Server via MRCP. > > Who wants to pitch in please paypal brian at freeswitch.org and I can > wire it to Arsen. > > Thanks, > Brian > > > Begin forwarded message: > >> ----- Forwarded Message ---- >> From: Arsen Chaloyan >> To: UniMRCP >> Sent: Wednesday, June 17, 2009 10:57:30 PM >> Subject: [UniMRCP] Open source ASR and TTS plugins >> >> Anybody interested in the development of open source ASR and TTS >> plugins for UniMRCP server write me offlist. >> >> PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ >> Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ >> >> Thanks, >> -- >> Arsen Chaloyan >> The author of UniMRCP >> http://www.unimrcp.org > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/8a83c0ef/attachment.html From brian at freeswitch.org Wed Jun 17 11:54:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:54:03 -0500 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia In-Reply-To: References: Message-ID: Its not possible right now but you could if you enable the config option and apply the tag... its something I have thought about adding but wasn't high on my list. NTATAG_SIPFLAGS(MSG_FLG_COMPACT) http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 /b On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: > Hi, > > Is it possible to enable compact SIP headers in mod_sofia > configuration? If yes, then how to do so? Kindly give an example. > > Thank you. From brian at freeswitch.org Wed Jun 17 11:55:19 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:55:19 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: mod_pocketsphinx will be there still as will mod_flite.. this lets you offload ASR and TTS to another server in a standard way. ScribleJ hasn't really helped me with mod_pocketsphinx. The out of grammar segfault is now gone if you update ;) /b On Jun 17, 2009, at 1:50 PM, Mitul Limbani wrote: > Brain, > > I can chip in $150 of my own as well, but two things: > > 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work > on it?) > 2) What are the deliverable we expecting of Arsen's effort? And a > generic ETA of those deliverables? > > Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt Ltd, > The Enterprise Linux Company(r), > http://www.enterux.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/45b407fe/attachment.html From edpimentl at gmail.com Wed Jun 17 11:56:31 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 17 Jun 2009 14:56:31 -0400 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> I will match the 150.00 Best regards, -E CEO and Founder Gpro.ws http://Twitter.com/edpimentl http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://TweetUp.ws (Twitter based MeetUp service) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/f88436a5/attachment.html From brian at freeswitch.org Wed Jun 17 11:59:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:59:57 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> Message-ID: <81ACBAA5-DCF6-4475-B225-F12A217D3BA4@freeswitch.org> Thanks for the match. /b On Jun 17, 2009, at 1:56 PM, EdPimentl wrote: > I will match the 150.00 > > Best regards, > -E > CEO and Founder > Gpro.ws > http://Twitter.com/edpimentl > > http://TwebEX.com (Twitter Based Online Web Conference Platform) > http://TwitrShare.com (Send Picture and Message to Tweet Contacts) > http://TweetUp.ws (Twitter based MeetUp service) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/2f4bb10e/attachment.html From kevin at johnnyvoip.com Wed Jun 17 11:57:29 2009 From: kevin at johnnyvoip.com (Kevin Green) Date: Wed, 17 Jun 2009 14:57:29 -0400 Subject: [Freeswitch-users] Solaris 10 - Install Instructions Needed Message-ID: I was wondering if anyone had a set of detailed step-by-step instructions on how to get FS up and running on Solaris 10 from a base Solaris install. I understand that there are a few pieces required to be able to build FS as described on the wiki ( http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris) but it skips over the details of each step. If someone could fill in the details or provide me with some help it would be greatly appreciated. Also, if there are any differences between working on an x86 vs a SPARC platform it would be good to have notes on the different steps that should be taken including any configuration variables that would optomize FS for x86 or SPARC CoolThreads machine. I have attached a whitepaper describing the benefit of UltraSPARC T2 processors running VoIP applications for anyone's interest. This whitepaper is my motivation for wanting to try FS on Solaris. Regards, Kevin Green JohnnyVoIP 350 Legget Drive Kanata, ON, Canada K2K 2W7 Phone: 613 271 5993 Ext 1203 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a6af22ae/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: 02a.SMI.323.SIP_UltraSPARC_WP.pdf Type: application/pdf Size: 145089 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a6af22ae/attachment-0001.pdf From brian at freeswitch.org Wed Jun 17 12:00:10 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 14:00:10 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: <04FB0300-BED1-4B8A-8F95-21D767B3B0C6@freeswitch.org> Btw thanks for the donation! ;) /b On Jun 17, 2009, at 1:50 PM, Mitul Limbani wrote: > Brain, > > I can chip in $150 of my own as well, but two things: > > 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work > on it?) > 2) What are the deliverable we expecting of Arsen's effort? And a > generic ETA of those deliverables? > > Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt Ltd, > The Enterprise Linux Company(r), > http://www.enterux.com/ > > > On 17-Jun-09, at 23:35, Brian West wrote: > >> Guys I'm tossing in $250 dollars of my own money on this ... who is >> going to pitch in? Arsen and I have been talking about how to >> accomplish this so we'll have an Open Source Speech Server via MRCP. >> >> Who wants to pitch in please paypal brian at freeswitch.org and I can >> wire it to Arsen. >> >> Thanks, >> Brian >> >> >> Begin forwarded message: >> >>> ----- Forwarded Message ---- >>> From: Arsen Chaloyan >>> To: UniMRCP >>> Sent: Wednesday, June 17, 2009 10:57:30 PM >>> Subject: [UniMRCP] Open source ASR and TTS plugins >>> >>> Anybody interested in the development of open source ASR and TTS >>> plugins for UniMRCP server write me offlist. >>> >>> PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ >>> Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ >>> >>> Thanks, >>> -- >>> Arsen Chaloyan >>> The author of UniMRCP >>> http://www.unimrcp.org >> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> dev >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/55351f8a/attachment.html From larclap at yahoo.com Wed Jun 17 12:09:26 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 17 Jun 2009 12:09:26 -0700 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? In-Reply-To: <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> References: <012901c9ef6e$01b3ee60$051bcb20$@com> <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> Message-ID: <019901c9ef7f$221498b0$663dca10$@com> Michael, The expression is part of version 13723 distribution in conf/dialplan/default.xml. Shouldn't that be changed? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 17, 2009 10:23 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] eavesdrop extension condition in default.xml? On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb wrote: In conf/dialplan/default.xml, the eavesdrop extension's condition is - expression="^88(.*)$|^\*0(.*)$"> Is this intended? I thought it was defined to eavesdrop on internal extensions. Why wouldn't it be something like - expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop extension. Lars, "8885551212" should never match regex expression "^88(\d{3,5})$" . It looks to me like your eavesdrop is matching on "^88(.*)$" which most definitely WILL match 8885551212. Please check your dialplan's eavesdrop extension's regex and make sure it is correct. -MC Log: 1030 Dialplan: sofia/internal/1009 at 192..168.10.29 parsing [default->eavesdrop] continue=false 1031 Dialplan: sofia/internal/1009 at 192..168.10.29 Regex (PASS) [eavesdrop] destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false 1032 Dialplan: sofia/internal/1009 at 192..168.10.29 Action answer() Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/16c1cd04/attachment.html From rupa at rupa.com Wed Jun 17 12:12:19 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 17 Jun 2009 14:12:19 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> Message-ID: I added the ability to call into your dialplan from the caller controls in conferences a while back. Depending on your goal, that might be an easy way to get your problem resolved. You can keep state using the hash api and hash on the conference name or some other useful thingie. On Wed, Jun 17, 2009 at 1:18 PM, Bradley Brashier wrote: > Well, since what I really need at this time is only about 5 commands of > similar complexity to a toggle on something already extant, I've decided to > just modify source. I can't imagine that people will be terribly interested > in my modifications, but I know I'm interested in being able to stay updated > with the current trunk, so I'll have to figure out how to deal with that. > > I'll let you know if I have trouble. > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/83cc02dd/attachment.html From shaheryarkh at googlemail.com Wed Jun 17 12:22:09 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 18 Jun 2009 01:22:09 +0600 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia In-Reply-To: References: Message-ID: Ok, thanks, i will take care of it in my code where necessary. Thank you. On Thu, Jun 18, 2009 at 12:54 AM, Brian West wrote: > Its not possible right now but you could if you enable the config > option and apply the tag... its something I have thought about adding > but wasn't high on my list. > > NTATAG_SIPFLAGS(MSG_FLG_COMPACT) > > > http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 > > /b > > On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: > > > Hi, > > > > Is it possible to enable compact SIP headers in mod_sofia > > configuration? If yes, then how to do so? Kindly give an example. > > > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/9f8f0481/attachment-0001.html From achaloyan at yahoo.com Wed Jun 17 12:56:29 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Wed, 17 Jun 2009 12:56:29 -0700 (PDT) Subject: [Freeswitch-users] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <61DF2924-3589-4DD5-9B2A-6CCF524DB3D2@freeswitch.org> References: <61DF2924-3589-4DD5-9B2A-6CCF524DB3D2@freeswitch.org> Message-ID: <750446.41188.qm@web111302.mail.gq1.yahoo.com> Mitul, >2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Both PocketSphinx and Flite will be integrated into open source UniMRCP server as ASR and TTS plugins respectively and will be publicly available under the Apache 2.0 license as UniMRCP itself. It's matter of 2~3 weeks to have basically working and usable plugins for them. Thank you, Arsen. From: Mitul Limbani Date: June 17, 2009 1:50:52 PM CDT To: "freeswitch-dev at lists.freeswitch.org" Cc: "freeswitch-dev at lists.freeswitch.org" , "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins Reply-To: freeswitch-users at lists.freeswitch.org Brain, I can chip in $150 of my own as well, but two things: 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work on it?) 2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 17-Jun-09, at 23:35, Brian West wrote: Guys I'm tossing in $250 dollars of my own money on this ... who is going to pitch in? Arsen and I have been talking about how to accomplish this so we'll have an Open Source Speech Server via MRCP. Who wants to pitch in please paypal brian at freeswitch.org and I can wire it to Arsen. Thanks, Brian Begin forwarded message: ----- Forwarded Message ---- From: Arsen Chaloyan To: UniMRCP Sent: Wednesday, June 17, 2009 10:57:30 PM Subject: [UniMRCP] Open source ASR and TTS plugins Anybody interested in the development of open source ASR and TTS plugins for UniMRCP server write me offlist. PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ Thanks, -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/154709ba/attachment.html From msc at freeswitch.org Wed Jun 17 13:08:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Jun 2009 15:08:15 -0500 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? In-Reply-To: <019901c9ef7f$221498b0$663dca10$@com> References: <012901c9ef6e$01b3ee60$051bcb20$@com> <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> <019901c9ef7f$221498b0$663dca10$@com> Message-ID: <87f2f3b90906171308g73499dccj9520caacdfd73593@mail.gmail.com> On Wed, Jun 17, 2009 at 2:09 PM, Lars Zeb wrote: > Michael, > > > > The expression is part of version 13723 distribution in > conf/dialplan/default.xml. Shouldn?t that be changed? > > > > > > "^88(.*)$|^\*0(.*)$"> > > > > "${hash(select/${domain_name}-spymap/$1)}"/> > > > > > > > Yes, it should be changed. What I'm saying is that the dialplan is matching on the regex expression "^88(.*)$" which will match your toll free numbers. In plain English it means that your dialplan is not correct somewhere. If you've got the new expression of "^88(\d{3,5})$" in your dialplan then that SHOULD work. Go double-check your dialplan and make sure that you don't have the wrong regex in there somewhere. Also, make sure that you reloadxml (or press F6) to make sure that your new changes are all properly loaded. -MC > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, June 17, 2009 10:23 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] eavesdrop extension condition in > default.xml? > > > > > > On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb wrote: > > In conf/dialplan/default.xml, the eavesdrop extension's condition is - > expression="^88(.*)$|^\*0(.*)$"> > > > > Is this intended? I thought it was defined to eavesdrop on internal > extensions. Why wouldn't it be something like - > expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial > 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop > extension. > > > > Lars, > "8885551212" should never match regex expression "^88(\d{3,5})$" . It > looks to me like your eavesdrop is matching on "^88(.*)$" which most > definitely WILL match 8885551212. Please check your dialplan's eavesdrop > extension's regex and make sure it is correct. > > -MC > > Log: > > 1030 Dialplan: sofia/internal/1009 at 192..168.10.29parsing [default->eavesdrop] continue=false > > 1031 Dialplan: sofia/internal/1009 at 192..168.10.29Regex (PASS) [eavesdrop] destination_number(8885819795) =~ > /^88(.*)$|^\*0(.*)$/ break=on-false > > 1032 Dialplan: sofia/internal/1009 at 192..168.10.29Action answer() > > > > Thanks, Lars > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/b1ebcf4d/attachment-0001.html From raul at etellicom.com Wed Jun 17 13:10:32 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 17 Jun 2009 17:10:32 -0300 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617073606.GA13623@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> Message-ID: <1245269432.16148.22.camel@raul-laptop> I can confirm the same issue, but it happens even with all the IPv6 stuff removed. This is my sofia status: freeswitch at internal> sofia status Name Type Data State ================================================================================================= internal profile sip:mod_sofia at 10.0.13.10:5060 RUNNING (0) imxtony.sytes.net alias internal ALIASED external profile sip:mod_sofia at 10.0.13.10:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG italk2 gateway sip:6499744069 at akl.italk.co.nz REGED italk gateway sip:6499744074 at akl.italk.co.nz REGED ================================================================================================= 2 profiles 1 alias And this is the INVITE for a gateway from an internal endpoint: freeswitch at internal> originate sofia/imxtony.sytes.net/218 &bridge(sofia/gateway/italk/6499744074) ------------------------------------------------------------------------ send 1344 bytes to udp/[203.184.16.2]:5060 at 20:01:50.190193: ------------------------------------------------------------------------ INVITE sip:6499744074 at akl.italk.co.nz SIP/2.0 Via: SIP/2.0/UDP 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np Max-Forwards: 70 From: "FreeSWITCH" ;tag=KSgjD3t33N9yp To: Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 CSeq: 116516120 INVITE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk- at SWITCH_VERSION_REVISION@ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Proxy-Authorization: Digest username="6499744074", realm="italk.co.nz", nonce="75444262", algorithm=MD5, uri="sip:6499744074 at akl.italk.co.nz", response="89bb5673f48252622025641153b882de" Content-Type: application/sdp Content-Disposition: session Content-Length: 315 Remote-Party-ID: "FreeSWITCH" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1245252480 1245252481 IN IP6 stun:stun.freeswitch.org s=FreeSWITCH c=IN IP6 stun:stun.freeswitch.org t=0 0 m=audio 16430 RTP/AVP 0 3 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2009-06-18 08:01:50.191067 [DEBUG] sofia.c:3210 Channel sofia/external/6499744074 entering state [calling][0] recv 469 bytes from udp/[203.184.16.2]:5060 at 20:01:50.252230: ------------------------------------------------------------------------ SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np;received=60.234.179.34 From: "FreeSWITCH" ;tag=KSgjD3t33N9yp To: ;tag=as34133b10 Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 CSeq: 116516120 INVITE User-Agent: italk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ------------------------------------------------------------------------ send 360 bytes to udp/[203.184.16.2]:5060 at 20:01:50.252988: ------------------------------------------------------------------------ ACK sip:6499744074 at akl.italk.co.nz SIP/2.0 Via: SIP/2.0/UDP 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np Max-Forwards: 70 From: "FreeSWITCH" ;tag=KSgjD3t33N9yp To: ;tag=as34133b10 Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 CSeq: 116516120 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-06-18 08:01:50.253252 [DEBUG] sofia.c:3210 Channel sofia/external/6499744074 entering state [terminated][488] That IP6 is clearly messing things up. Regards, Raul On Wed, 2009-06-17 at 17:36 +1000, Jason White wrote: > Jason White wrote: > > > The symptom is the following line in outgoing SIP messages while attempting to > > establish a call to a gateway via the external profile: > > > > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org > > However, if I place an IPv6 call via the internal-ipv6 profile, the o= line > contains the correct IPv6 address - so this is only adversely affecting the > IPv4 external profile, it seems. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/f98483e0/attachment.html From jmesquita at gmail.com Wed Jun 17 13:26:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 17 Jun 2009 17:26:50 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <24067617.post@talk.nabble.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> Message-ID: <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP "tricks" 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: > > Hi, > > look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" > where you can ask for special requirements. > > NOx > > > > Diego Viola wrote: > > > > Hi everyone, > > > > Can you please recommend me some GSM gateway? I'm currently looking > > for a good one to buy... anyone have experience PORTech GSM gateways? > > Are they good? > > > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > > > Thanks, > > > > Diego > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/4a2d570a/attachment.html From apt.get at gmail.com Wed Jun 17 14:15:43 2009 From: apt.get at gmail.com (David Burgess) Date: Wed, 17 Jun 2009 15:15:43 -0600 Subject: [Freeswitch-users] high latency In-Reply-To: References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> Message-ID: On Fri, Jun 12, 2009 at 5:34 PM, Mathieu Rene wrote: > Try > Math This seems to have greatly reduced but not eliminated late-onset latency :p The latest version of the pfsense-freeswitch package is based on freeswitch 13784 and claims to have eliminated the issue, but I haven't tried it yet. db From jan.kubr at gmail.com Wed Jun 17 14:31:06 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Wed, 17 Jun 2009 23:31:06 +0200 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> Message-ID: <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan 2009/6/17 Jo?o Mesquita > Guys, I was looking at the advantages and disadvantages of having a GSM > gateway vs. a GSM board. > > The conclusions I get are: > > Board pros > > 1. Boards are able to get/send SMS without SIP "tricks" > 2. You don't have to make a SIP call to check if channel is available and > don't rely o SIP messages to get channel status > 3. FS will be able to check for signal level on the board and fire events > on pre-defined thresholds. > > Gateway pros > > 1. I think of is the a GW can be used by more then one server, therefore, > can have failover. > 2. A GW is more scalable > > It would be nice if you, that have already used GSM GWs in production, > could comment on this. > > Thanks, > > jmesquita > > > On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: > >> >> Hi, >> >> look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" >> where you can ask for special requirements. >> >> NOx >> >> >> >> Diego Viola wrote: >> > >> > Hi everyone, >> > >> > Can you please recommend me some GSM gateway? I'm currently looking >> > for a good one to buy... anyone have experience PORTech GSM gateways? >> > Are they good? >> > >> > I also need it to work with FS, I'm also kinda new with VoIP hardware. >> > >> > Thanks, >> > >> > Diego >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/4b731506/attachment-0001.html From jmesquita at gmail.com Wed Jun 17 14:43:46 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 17 Jun 2009 18:43:46 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> Message-ID: <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> Pricewise, is it worth it? jmesquita On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: > We plan to buy one of these: > > http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html > since you can use SMTP/POP3 to manage SMS. > > Jan > > 2009/6/17 Jo?o Mesquita > > Guys, I was looking at the advantages and disadvantages of having a GSM >> gateway vs. a GSM board. >> >> The conclusions I get are: >> >> Board pros >> >> 1. Boards are able to get/send SMS without SIP "tricks" >> 2. You don't have to make a SIP call to check if channel is available and >> don't rely o SIP messages to get channel status >> 3. FS will be able to check for signal level on the board and fire events >> on pre-defined thresholds. >> >> Gateway pros >> >> 1. I think of is the a GW can be used by more then one server, therefore, >> can have failover. >> 2. A GW is more scalable >> >> It would be nice if you, that have already used GSM GWs in production, >> could comment on this. >> >> Thanks, >> >> jmesquita >> >> >> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >> >>> >>> Hi, >>> >>> look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" >>> where you can ask for special requirements. >>> >>> NOx >>> >>> >>> >>> Diego Viola wrote: >>> > >>> > Hi everyone, >>> > >>> > Can you please recommend me some GSM gateway? I'm currently looking >>> > for a good one to buy... anyone have experience PORTech GSM gateways? >>> > Are they good? >>> > >>> > I also need it to work with FS, I'm also kinda new with VoIP hardware. >>> > >>> > Thanks, >>> > >>> > Diego >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a307bda5/attachment.html From R.Kloosterman at mtel.nl Wed Jun 17 14:44:25 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Wed, 17 Jun 2009 23:44:25 +0200 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com><24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A9AF@srvmtel.office.mtel.nl> I have used board-based voice solutions for ISDN/SS7 over the last 10 years and just moved to gateways and VoIP 2 years ago. I must say I've never been more happy. It scales excellent and is very stable, never had any problem and it's handling about 30.000 calls per day. In the past I've had a fair share of board issues, some crashes, scaling headache. Not totaly bad, but the gateway approach is much nicer. Check out audiocodes, I believe their mediant series supports GSM as well. Remko ________________________________ Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Jo?o Mesquita Verzonden: woensdag 17 juni 2009 22:27 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] Which GSM gateway to buy? Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP "tricks" 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: Hi, look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" where you can ask for special requirements. NOx Diego Viola wrote: > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a612bce2/attachment.html From bjbrashier at gmail.com Wed Jun 17 14:45:52 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Wed, 17 Jun 2009 14:45:52 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> Message-ID: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> So I found one interesting thing so far: the "lock" caller control actually does function as a toggle, and, in fact, "unlock" does not do anything. This goes against wiki docs on mod_conference, but is helpful in this instance. I have a few other commands to work on, still. I found execute-application to be interesting, but since what I next need is a count of all conference participants, and no application already exists for that, I'm still going to have to write something else. BB On Wed, Jun 17, 2009 at 12:12 PM, Rupa Schomaker wrote: > I added the ability to call into your dialplan from the caller controls in > conferences a while back. Depending on your goal, that might be an easy way > to get your problem resolved. You can keep state using the hash api and > hash on the conference name or some other useful thingie. > > On Wed, Jun 17, 2009 at 1:18 PM, Bradley Brashier wrote: > >> Well, since what I really need at this time is only about 5 commands of >> similar complexity to a toggle on something already extant, I've decided to >> just modify source. I can't imagine that people will be terribly interested >> in my modifications, but I know I'm interested in being able to stay updated >> with the current trunk, so I'll have to figure out how to deal with that. >> >> I'll let you know if I have trouble. >> > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/29535a5a/attachment.html From steve at justfone.com Wed Jun 17 14:53:26 2009 From: steve at justfone.com (Steven Brown) Date: Wed, 17 Jun 2009 22:53:26 +0100 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 172 In-Reply-To: References: Message-ID: <3e6d7b0c0906171453y149cbb0n6a4d3522a7495606@mail.gmail.com> You can also send and recieve SMS on the PORTech Gateways via AT commands over a socket connection, I've not checked but would assume all other AT commands will work so you can check signal strength etc this way also. Steve Message: 4 Date: Wed, 17 Jun 2009 23:31:06 +0200 From: Jan Kubr Subject: Re: [Freeswitch-users] Which GSM gateway to buy? To: freeswitch-users at lists.freeswitch.org Message-ID: <698401620906171431t10432015xa78976a401dc5c5b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/601ffc4c/attachment-0001.html From jaybinks at gmail.com Wed Jun 17 15:28:29 2009 From: jaybinks at gmail.com (jay binks) Date: Thu, 18 Jun 2009 08:28:29 +1000 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> Message-ID: Ive used these in the past. http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html sound fine, work well... reliable etc etc.. things to watch out for... : * cant send your own caller ID from them ( in my experience its locked to the sim ) * your provider might block the IMEI number of the GSM terminal, if they dont like what your doing. just some stuff to consider. Jay 2009/6/18 Jo?o Mesquita > Pricewise, is it worth it? > > jmesquita > > > On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: > >> We plan to buy one of these: >> >> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >> since you can use SMTP/POP3 to manage SMS. >> >> Jan >> >> 2009/6/17 Jo?o Mesquita >> >> Guys, I was looking at the advantages and disadvantages of having a GSM >>> gateway vs. a GSM board. >>> >>> The conclusions I get are: >>> >>> Board pros >>> >>> 1. Boards are able to get/send SMS without SIP "tricks" >>> 2. You don't have to make a SIP call to check if channel is available and >>> don't rely o SIP messages to get channel status >>> 3. FS will be able to check for signal level on the board and fire events >>> on pre-defined thresholds. >>> >>> Gateway pros >>> >>> 1. I think of is the a GW can be used by more then one server, therefore, >>> can have failover. >>> 2. A GW is more scalable >>> >>> It would be nice if you, that have already used GSM GWs in production, >>> could comment on this. >>> >>> Thanks, >>> >>> jmesquita >>> >>> >>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >>> >>>> >>>> Hi, >>>> >>>> look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" >>>> where you can ask for special requirements. >>>> >>>> NOx >>>> >>>> >>>> >>>> Diego Viola wrote: >>>> > >>>> > Hi everyone, >>>> > >>>> > Can you please recommend me some GSM gateway? I'm currently looking >>>> > for a good one to buy... anyone have experience PORTech GSM gateways? >>>> > Are they good? >>>> > >>>> > I also need it to work with FS, I'm also kinda new with VoIP hardware. >>>> > >>>> > Thanks, >>>> > >>>> > Diego >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/b5eec936/attachment.html From jason at jasonjgw.net Wed Jun 17 18:01:30 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 18 Jun 2009 11:01:30 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <1245269432.16148.22.camel@raul-laptop> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> Message-ID: <20090618010130.GA14634@jdc.jasonjgw.net> Raul Fragoso wrote: > I can confirm the same issue, but it happens even with all the IPv6 > stuff removed. Thank you for the corroboration. It only happens to me if I have the following in my external.xml profile: Note that I need the above line for nat traversal; if I leave it out, FreeSWITCH uses a private IPv4 address in outgoing invites, and can't establish a call. From jingwei.yang at gmail.com Wed Jun 17 19:50:35 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 18 Jun 2009 10:50:35 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <191c3a030906170618v214fb0e9y4d5c64b724a97200@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> <191c3a030906170618v214fb0e9y4d5c64b724a97200@mail.gmail.com> Message-ID: <13529f9d0906171950j26df5790r19e0ccd8328dcd4d@mail.gmail.com> Hi Anthony, I've filed the report earlier on. This is the url of the bug: http://jira.freeswitch.org/browse/MODSKYPIAX-35. Hi Giovanni, Since the work is still under the development phase, I only managed to get 8 concurrent calls. Here's the cpu and memory consumption data: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 3822 root 15 0 123m 39m 10m S 8.6 1.0 0:37.60 skype 4801 root 18 0 48304 20m 5352 S 4.0 0.5 0:12.24 freeswitch 3862 root 15 0 122m 37m 10m S 3.3 0.9 0:18.73 skype 3842 root 15 0 120m 36m 10m S 2.3 0.9 0:26.36 skype 3882 root 15 0 121m 37m 10m S 2.3 0.9 0:20.63 skype 3922 root 15 0 98.0m 33m 9604 S 2.3 0.8 0:12.77 skype 3902 root 15 0 120m 36m 10m S 2.0 0.9 0:15.89 skype 3942 root 15 0 99.9m 34m 9628 S 1.3 0.9 0:12.05 skype 4065 root 15 0 71272 33m 9180 S 0.7 0.8 0:06.50 skype 4100 root 15 0 14060 8708 1804 S 0.7 0.2 0:04.55 Xvfb Regards, -Jingwei On Wed, Jun 17, 2009 at 9:18 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You should not report bugs on the mailing list. > please report your bug to jira http://jira.freeswitch.org > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Make sure you attach a backtrace of your issue and file it under skypeiax > so giovanni can track it. > > > On Wed, Jun 17, 2009 at 4:39 AM, Jingwei Yang wrote: > >> Hi Giovanni, >> >> Sorry, pretty busy and fully occupied by other stuff today. Have to delay >> the testing and give you the result tomorrow. >> >> Regards, >> -Jingwei >> >> >> On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang wrote: >> >>> Sure, I'll append to you the result tomorrow. >>> >>> Regards, >>> -Jingwei >>> >>> >>> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli < >>> gmaruzz at celliax.org> wrote: >>> >>>> Hi Jingwei, >>>> >>>> Thanks a lot! I'll take care of as soon as possible. >>>> >>>> Btw, before I read the Jira, are you testing in linux? >>>> >>>> If you are testing on linux, would you please report how it is >>>> performing under load? I mean, what is the load average with, let say, >>>> 10 or 20 or more concurrent Skype call? >>>> >>>> This has nothing to do with your bug, but will help me in getting >>>> better performances. >>>> >>>> Ciao for now, and thanks again for reporting! >>>> >>>> -giovanni >>>> >>>> >>>> >>>> >>>> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >>>> wrote: >>>> > Hi Giovanni, >>>> > >>>> > I've reported it in Jira. Here's the bug url: >>>> > >>>> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >>>> > >>>> > Thanks, >>>> > -Jingwei >>>> > >>>> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < >>>> gmaruzz at celliax.org> >>>> > wrote: >>>> >> >>>> >> Hi Jingwel, >>>> >> thanks for reporting. >>>> >> >>>> >> Could you please add a Jira issue with as much details as possible? >>>> >> >>>> >> general guide for reporting bugs: >>>> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >> >>>> >> what to add for skypiax: >>>> >> >>>> >> >>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >>>> >> >>>> >> mod_skypiax Jira: >>>> >> http://jira.freeswitch.org/browse/MODSKYPIAX >>>> >> >>>> >> >>>> >> Sincerely, >>>> >> >>>> >> Giovanni Maruzzelli >>>> >> ========================================= >>>> >> www.celliax.org >>>> >> via Pierlombardo 9, 20135 Milano >>>> >> Italy >>>> >> gmaruzz at celliax dot org >>>> >> Cell : +39-347-2665618 >>>> >> Fax : +39-02-87390039 >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang< >>>> jingwei.yang at gmail.com> >>>> >> wrote: >>>> >> > Hi Team, >>>> >> > >>>> >> > I've been using the record_session feature to record call sessions. >>>> >> > Here's >>>> >> > how I prepared the dialplan: >>>> >> > >>>> >> > >>>> >> > >>> expression="^2909/(.*)$"> >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > And here's how I trigger it: >>>> >> > >>>> >> > freeswitch at localhost.localdomain>originate >>>> skypiax/skypiax2/userAAA >>>> >> > 2909/userBBB >>>> >> > >>>> >> > The call can be established and the data.wav file was generated >>>> without >>>> >> > any >>>> >> > problem. However, once userAAA hung up, a segmentation fault >>>> occurred >>>> >> > and >>>> >> > freeswitch was automatically shut down. Here are what I got from >>>> the >>>> >> > console: >>>> >> > >>>> >> > freeswitch at localhost.localdomain> originate >>>> skypiax/skypiax2/userAAA >>>> >> > 2909/userBBB >>>> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >>>> >> > switch_channel_set_name() >>>> >> > New Channel skypiax/skypiax2/userAAA >>>> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >>>> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >>>> >> > remote_party_is_ringing() >>>> >> > Ring-Ready skypiax/skypiax2/userAAA >>>> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >>>> >> > outbound_channel_answered() >>>> >> > Channel [skypiax/skypiax2/userAAA] has been answered >>>> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >>>> >> > switch_ivr_session_transfer() >>>> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >>>> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >>>> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >>>> >> > >>>> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >>>> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >>>> >> > FreeSWITCH->2909/userBBB >>>> >> > in context default >>>> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >>>> >> > switch_channel_set_name() >>>> >> > New Channel skypiax/ANY/userBBB >>>> [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >>>> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >>>> >> > remote_party_is_ringing() >>>> >> > Ring-Ready skypiax/ANY/userBBB! >>>> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >>>> >> > outbound_channel_answered() >>>> >> > Channel [skypiax/ANY/userBBB] has been answered >>>> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >>>> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >>>> >> > [CS_EXECUTE] >>>> >> > [NORMAL_CLEARING] >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >>>> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >>>> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>>> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >>>> Ended >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>>> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >>>> >> > [CS_DESTROY] >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>>> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>>> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >>>> >> > [CS_DESTROY] >>>> >> > Segmentation fault (core dumped) >>>> >> > >>>> >> > Please kindly let me know whether there's anything wrong with the >>>> >> > dialplan >>>> >> > or the way how I originated the call. >>>> >> > >>>> >> > Thanks! >>>> >> > -Jingwei >>>> >> > >>>> >> > >>>> >> > _______________________________________________ >>>> >> > Freeswitch-users mailing list >>>> >> > Freeswitch-users at lists.freeswitch.org >>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> > http://www.freeswitch.org >>>> >> > >>>> >> > >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/1a435bd5/attachment-0001.html From j3flight at gmail.com Wed Jun 17 17:13:56 2009 From: j3flight at gmail.com (j3flight) Date: Wed, 17 Jun 2009 17:13:56 -0700 (PDT) Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> Message-ID: <24084409.post@talk.nabble.com> I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... Here's a wiki page I created after building a JavaScript IVR for a conference server... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I believe you could stick those in a script by themselves and call them from execute_application. Somehow, you would have to identify what user is calling the script and what conference they're in. (You could possibly set a session variable upon entering the conference, or parse all the conferences until you find that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! -- View this message in context: http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24084409.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From j3flight at gmail.com Wed Jun 17 17:51:13 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Wed, 17 Jun 2009 19:51:13 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> Message-ID: <4A398F81.1020303@gmail.com> I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... Here's a wiki page I created after building a JavaScript IVR for a conference server... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I believe you could stick those in a script by themselves and call them from execute_application. Somehow, you would have to identify what user is calling the script and what conference they're in. (Once you're in the javascript, you could check a "conference number" variable that you set when the person entered the conference. Or, you could parse the output of "conference list" until you found that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! From jcromes at gmail.com Wed Jun 17 20:10:41 2009 From: jcromes at gmail.com (Jason Cromes) Date: Wed, 17 Jun 2009 22:10:41 -0500 Subject: [Freeswitch-users] Controlling Conference Controls Message-ID: I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... This Wiki page has some good JavaScript examples... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I *believe* you could stick those in a script by themselves and call them using execute_application out of the caller controls... Somehow, you would have to identify what user is calling the script and what conference they're in.? (Once you're in the javascript, you could check a "conference number" variable that you set when the person entered the conference.? Or, you could parse the output of "conference list" until you found that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! From j3flight at gmail.com Wed Jun 17 21:07:28 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Wed, 17 Jun 2009 23:07:28 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> Message-ID: <4A39BD80.8020706@gmail.com> FYI: I fixed the Wiki documentation for the lock/unlock feature. Bradley Brashier wrote: > So I found one interesting thing so far: the "lock" caller control > actually does function as a toggle, and, in fact, "unlock" does not do > anything. This goes against wiki docs on mod_conference, but is > helpful in this instance. From brian at freeswitch.org Wed Jun 17 21:57:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 23:57:03 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090618010130.GA14634@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> <20090618010130.GA14634@jdc.jasonjgw.net> Message-ID: <8B621D34-D0D9-4AAE-B010-36B56DFBBE4C@freeswitch.org> I need in one of your boxs... there is no way this is doing this unless you are putting stun:stun.freeswitch.org into the ext-rtp-ip or sip ip... which could make it trigger the ipv6 check since its just looking for : in the ip address. And stun: has that.. so you're triggering it... tar up your entire conf folder and mail it to me ASAP. /b On Jun 17, 2009, at 8:01 PM, Jason White wrote: > Raul Fragoso wrote: >> I can confirm the same issue, but it happens even with all the IPv6 >> stuff removed. > > Thank you for the corroboration. > > It only happens to me if I have the following in my external.xml > profile: > > > Note that I need the above line for nat traversal; if I leave it out, > FreeSWITCH uses a private IPv4 address in outgoing invites, and can't > establish a call. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Jun 17 22:04:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 00:04:16 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <1245269432.16148.22.camel@raul-laptop> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> Message-ID: <8D2072FB-72BA-4FF2-88E1-C80E200DAD63@freeswitch.org> Ok you both didn't notice you CAN NOT put stun:stun.freeswitch.org in rtp-ip, thats the problem. It clearly says IP ADDRESSES ONLY in the comments. DO not use $${external_rtp_ip} for rtp-ip either :P /b On Jun 17, 2009, at 3:10 PM, Raul Fragoso wrote: > I can confirm the same issue, but it happens even with all the IPv6 > stuff removed. > This is my sofia status: > > freeswitch at internal> sofia status > > Name Type Data > State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > internal profile sip:mod_sofia at 10.0.13.10:5060 > RUNNING (0) > imxtony.sytes.net alias internal > ALIASED > external profile sip:mod_sofia at 10.0.13.10:5080 > RUNNING (0) > example.com gateway sip:joeuser at example.com > NOREG > italk2 gateway sip:6499744069 at akl.italk.co.nz > REGED > italk gateway sip:6499744074 at akl.italk.co.nz > REGED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 2 profiles 1 alias > > And this is the INVITE for a gateway from an internal endpoint: > > freeswitch at internal> originate sofia/imxtony.sytes.net/218 > &bridge(sofia/gateway/italk/6499744074) > > ------------------------------------------------------------------------ > send 1344 bytes to udp/[203.184.16.2]:5060 at 20:01:50.190193: > > ------------------------------------------------------------------------ > INVITE sip:6499744074 at akl.italk.co.nz SIP/2.0 > Via: SIP/2.0/UDP > 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np > Max-Forwards: 70 > From: "FreeSWITCH" 6499744074 at akl.italk.co.nz;transport=udp>;tag=KSgjD3t33N9yp > To: > Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 > CSeq: 116516120 INVITE > Contact: > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk- > @SWITCH_VERSION_REVISION@ > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Proxy-Authorization: Digest username="6499744074", > realm="italk.co.nz", nonce="75444262", algorithm=MD5, uri="sip:6499744074 at akl.italk.co.nz > ", response="89bb5673f48252622025641153b882de" > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 315 > Remote-Party-ID: "FreeSWITCH" 0000000000 at akl.italk.co.nz>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1245252480 1245252481 IN IP6 stun:stun.freeswitch.org > s=FreeSWITCH > c=IN IP6 stun:stun.freeswitch.org > t=0 0 > m=audio 16430 RTP/AVP 0 3 8 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > 2009-06-18 08:01:50.191067 [DEBUG] sofia.c:3210 Channel sofia/ > external/6499744074 entering state [calling][0] > recv 469 bytes from udp/[203.184.16.2]:5060 at 20:01:50.252230: > > ------------------------------------------------------------------------ > SIP/2.0 488 Not acceptable here > Via: SIP/2.0/UDP > 60.234.179.34 > :5080;rport;branch=z9hG4bKgypXrUy0889Np;received=60.234.179.34 > From: "FreeSWITCH" 6499744074 at akl.italk.co.nz;transport=udp>;tag=KSgjD3t33N9yp > To: ;tag=as34133b10 > Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 > CSeq: 116516120 INVITE > User-Agent: italk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 360 bytes to udp/[203.184.16.2]:5060 at 20:01:50.252988: > > ------------------------------------------------------------------------ > ACK sip:6499744074 at akl.italk.co.nz SIP/2.0 > Via: SIP/2.0/UDP > 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np > Max-Forwards: 70 > From: "FreeSWITCH" 6499744074 at akl.italk.co.nz;transport=udp>;tag=KSgjD3t33N9yp > To: ;tag=as34133b10 > Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 > CSeq: 116516120 ACK > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2009-06-18 08:01:50.253252 [DEBUG] sofia.c:3210 Channel sofia/ > external/6499744074 entering state [terminated][488] > > > That IP6 is clearly messing things up. > > Regards, > > Raul > > On Wed, 2009-06-17 at 17:36 +1000, Jason White wrote: >> >> Jason White wrote: >> >> > The symptom is the following line in outgoing SIP messages while >> attempting to >> > establish a call to a gateway via the external profile: >> > >> > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org >> >> However, if I place an IPv6 call via the internal-ipv6 profile, the >> o= line >> contains the correct IPv6 address - so this is only adversely >> affecting the >> IPv4 external profile, it seems. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5b552778/attachment-0001.html From brian at freeswitch.org Wed Jun 17 22:09:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 00:09:44 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <1245269432.16148.22.camel@raul-laptop> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> Message-ID: <93A6FD81-394C-492A-894D-49C8D4CCB1AB@freeswitch.org> I no longer need your configs... I didn't try to put stun:stun.freeswitch.org in sip-ip or rtp-ip because I know you shouldn't. We clearly can not try to do a stun request in either of these fields because you can't bind to IP's that aren't directly on the machine... so do as per the config and put ONLY ip's that are local on the machine... if you need to set the external rtp or sip ip please set them with the ext-rtp-ip and ext-sip-ip params on the profile. You CAN set the ext-rtp-ip and ext-sip-ip to stun:stun.freeswitch.org and those will resolve. /b On Jun 17, 2009, at 3:10 PM, Raul Fragoso wrote: > I can confirm the same issue, but it happens even with all the IPv6 > stuff removed. > This is my sofia status: From darklion11 at yahoo.com Wed Jun 17 22:10:43 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 17 Jun 2009 22:10:43 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> Message-ID: <24086477.post@talk.nabble.com> Not working... CALL Rejected dujinfang wrote: > > comment lines in the user directory do the trick: > > > > > > > > On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote: > >> >> If FS A has an account 8011105 does FS B also nid to register >> 8011105? Yes it >> working on a gateway but the username of the gateway was shown on my >> softphone and also it nids a password for the gateway... is there an >> option >> to view the caller name and number of the FS A gateway to FS B? >> >> >> >> >> Brian West-3 wrote: >>> >>> COPY paste fail :) >>> >>> >>> >>> something like that as per the example. >>> >>> /b >>> >>> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >>> >>>> >>>> How can sofia profile can call ACL? >>>> Can you give me an example? >>>> Like this? >>>> >>>> I put this on external profile >>>> >>>> "/> >>>> "/> >>>> >>>> >>>> Brian West-3 wrote: >>>>> >>>>> Now you have to tell the sofia profile to use that ACL >>>>> >>>>> /b >>>>> >>>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>>> >>>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>>> 192.168.0.105 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 192.168.0.4 >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24086477.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Jun 17 22:14:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 00:14:12 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24086477.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24086477.post@talk.nabble.com> Message-ID: You're trying way too hard. CALL Rejected gives us exactly ZERO to go on... We are all trying really hard to help you but at some point we just can't help anymore. Please make sure you post debug logs to pastebin and join IRC. This email back and forth over something like this just takes way too long and frustrates everyone... frustration leads to trying to hard... which leads to failure. /b On Jun 18, 2009, at 12:10 AM, Edmar Cruz wrote: > > Not working... CALL Rejected From christian.bourke1 at gmail.com Wed Jun 17 23:03:21 2009 From: christian.bourke1 at gmail.com (Bilbo) Date: Wed, 17 Jun 2009 23:03:21 -0700 (PDT) Subject: [Freeswitch-users] Use Freeswitch to provide a SIP Video/Voice email service Message-ID: <24086865.post@talk.nabble.com> Hi, I would like to use Freeswitch to provide a Video/Voice mail service that is integrated with an email service. I would like to have the ability to email the Video/Voice messages as well as the SIP users being able to collect their Video messages using their video soft-phones. Has anyone done this before or know if Freeswitch is capable? Thanks -- View this message in context: http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-service-tp24086865p24086865.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Wed Jun 17 23:54:29 2009 From: dujinfang at gmail.com (seven) Date: Thu, 18 Jun 2009 14:54:29 +0800 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24086477.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24086477.post@talk.nabble.com> Message-ID: Note I was saying your caller id problem, how did you see the undesired caller id when you got CALL Rejected? On Jun 18, 2009, at 1:10 PM, Edmar Cruz wrote: > > Not working... CALL Rejected > > dujinfang wrote: >> >> comment lines in the user directory do the trick: >> >> >> >> >> >> >> >> On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote: >> >>> >>> If FS A has an account 8011105 does FS B also nid to register >>> 8011105? Yes it >>> working on a gateway but the username of the gateway was shown on my >>> softphone and also it nids a password for the gateway... is there an >>> option >>> to view the caller name and number of the FS A gateway to FS B? >>> >>> >>> From peter.olsson at visionutveckling.se Thu Jun 18 00:07:22 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 18 Jun 2009 09:07:22 +0200 Subject: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a "api uuid_record start ", and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/d07afbd7/attachment.html From rdenert at tng.de Thu Jun 18 00:17:29 2009 From: rdenert at tng.de (Rudolf Denert) Date: Thu, 18 Jun 2009 09:17:29 +0200 (CEST) Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <18011901.31411245309252614.JavaMail.root@zimbra.tng.de> Message-ID: <2557047.31431245309449504.JavaMail.root@zimbra.tng.de> Yes, I removed the tags but with no effect. I think the problem is that the webserver doesn't look in the directory where featuers.xml is deposited on the freeswitchserver (/opt/freeswitch/conf/dialplan/). The issue is that FS finds the context when dialplan is the directory /opt/freeswitch/conf/dialplan/public/ . But when it is on the webserver (it is on another server with a different IP-address) I get the error that I told you. What should i verify in the default config. Greetz ----- Urspr?ngliche Mail ----- Von: "Brian West" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Freeswitch / Webserver Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > Context features not found _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From dujinfang at gmail.com Thu Jun 18 00:20:22 2009 From: dujinfang at gmail.com (seven) Date: Thu, 18 Jun 2009 15:20:22 +0800 Subject: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> Message-ID: <7E13F947-F283-44AD-B718-429632E8319D@gmail.com> can you try uuid_record stop before playback? On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote: > I?m not quite sure if this is the expected behaviour, I just wanted > to make sure. > > I?ve developed a simple IVR application using event socket. I dial > in to the dialplan and park the call, and then I let the IVR > application do whatever it?s supposed to. I basically listen for > DTMF events and play and record files. > > Today I just noticed that if I issue a ?api uuid_record start > ?, and then do a file playback (using SendMsg, with call- > command execute and execute-app-name playback), the playback is sent > both to the caller, and to the recorded file. Is this the way it?s > supposed to work, or should I playback files in another way? > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/88bc875b/attachment.html From d at d-man.org Thu Jun 18 00:24:20 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 18 Jun 2009 00:24:20 -0700 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: <24065638.post@talk.nabble.com> References: <24065638.post@talk.nabble.com> Message-ID: Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn of "tcapi" still in the config file. If your test was: # isql zenoss edmar edmar Then zenoss should be your db_dsn: Not You should be seeing something about the ODBC connection failing at FreeSWITCH startup if you look at the log closely (search for mod_nibblebill) that indicates this, too. - Darren -----Original Message----- From: Edmar Cruz [mailto:darklion11 at yahoo.com] Sent: Tuesday, June 16, 2009 6:44 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC my nibble.conf.xml param name="db_username" value="edmar"/> param name="db_password" value="edmar"/> param name="db_dsn" value="tcapi"/> param name="db_column_cash" value="cash"/> param name="db_column_account" value="id"/> param name="global_heartbeat" value="1"/> !-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. --> param name="lowbal_amt" value="5"/> param name="lowbal_action" value="play ding"/> param name="nobal_amt" value="0"/> param name="nobal_action" value="hangup"/> param name="percall_max_amt" value="100"/> param name="percall_action" value="hangup"/> Account 1001.xml param name="password" value="1234"/> param name="vm-password" value="1001"/> param name="vm-mailto" value=""/> param name="vm-email-all-messages" value="false"/> param name="vm-delete-file" value="false"/> param name="vm-attach-file" value="false"/> I check unixodbc has been installed. # isql zenoss edmar edmar [SQL]> Connected successfully but on freeswitch error Cannot connect to user ODBC [root] Darren Schreiber wrote: > > What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the > real logs from FS's logs? The info below is not nearly detailed enough. > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Monday, June 15, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC > > > Hi > > I experiencing an error on mod_nibblebill. I already load it from > autoload_configs, especially mod_spidermonkey. Uncomment > mod_spidermonkey_odbc. I also download unixodbc and created the files > /etc/odbcinst.ini and /etc/odbc.ini with the correct format > > [zenoss] > DATABASE = tcapi > USER = root > PASS = password > ..... > > I type also on the console isql zenoss root password. Also working... > > But an error occur on freeswitch Cannot connect to user [root] ... > > What do you thinks is the problem? > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24045890.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 890p24065638.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From peter.olsson at visionutveckling.se Thu Jun 18 00:39:35 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 18 Jun 2009 09:39:35 +0200 Subject: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? In-Reply-To: <7E13F947-F283-44AD-B718-429632E8319D@gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> <7E13F947-F283-44AD-B718-429632E8319D@gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB947B@cooper> Yes I guess this would probably solve the issue :) But since I stumbled across this weird behaviour I just wanted to make sure if this was expected or not, or if it might be a bug... I thought playback was just sending the audio to the caller, but in this case it seems that playback sends it to both "parties". /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r seven Skickat: den 18 juni 2009 09:20 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? can you try uuid_record stop before playback? On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote: I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a "api uuid_record start ", and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a39ebaa32936831919445! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/e9b8f6a3/attachment.html From darklion11 at yahoo.com Thu Jun 18 00:58:36 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 18 Jun 2009 00:58:36 -0700 (PDT) Subject: [Freeswitch-users] Is freeswitch can call mobile phones? Message-ID: <24088222.post@talk.nabble.com> Hi is there any possible free sites ip that i can connect so I can could to any mobiles phones? I know some several ip sites has the capability to call for free Ip to Voip... I know freeswitch can do this Can you give me an example site? -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jan.kubr at gmail.com Thu Jun 18 01:26:26 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Thu, 18 Jun 2009 10:26:26 +0200 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> Message-ID: <698401620906180126n454589a7y3bc2ff4ae7004741@mail.gmail.com> There are gateways that allow you to set your own caller ID? I thought it'd always use the number of the SIM. Jan On Thu, Jun 18, 2009 at 12:28 AM, jay binks wrote: > Ive used these in the past. > > http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html > > sound fine, work well... > reliable etc etc.. > > things to watch out for... : > > * cant send your own caller ID from them ( in my experience its locked to > the sim ) > * your provider might block the IMEI number of the GSM terminal, if they > dont like what your doing. > > just some stuff to consider. > > > Jay > > > > > 2009/6/18 Jo?o Mesquita > > Pricewise, is it worth it? >> >> jmesquita >> >> >> On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: >> >>> We plan to buy one of these: >>> >>> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >>> since you can use SMTP/POP3 to manage SMS. >>> >>> Jan >>> >>> 2009/6/17 Jo?o Mesquita >>> >>> Guys, I was looking at the advantages and disadvantages of having a GSM >>>> gateway vs. a GSM board. >>>> >>>> The conclusions I get are: >>>> >>>> Board pros >>>> >>>> 1. Boards are able to get/send SMS without SIP "tricks" >>>> 2. You don't have to make a SIP call to check if channel is available >>>> and don't rely o SIP messages to get channel status >>>> 3. FS will be able to check for signal level on the board and fire >>>> events on pre-defined thresholds. >>>> >>>> Gateway pros >>>> >>>> 1. I think of is the a GW can be used by more then one server, >>>> therefore, can have failover. >>>> 2. A GW is more scalable >>>> >>>> It would be nice if you, that have already used GSM GWs in production, >>>> could comment on this. >>>> >>>> Thanks, >>>> >>>> jmesquita >>>> >>>> >>>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >>>> >>>>> >>>>> Hi, >>>>> >>>>> look at www.kuhnt.com. It?s a german page. There you can find >>>>> "Kontakt" >>>>> where you can ask for special requirements. >>>>> >>>>> NOx >>>>> >>>>> >>>>> >>>>> Diego Viola wrote: >>>>> > >>>>> > Hi everyone, >>>>> > >>>>> > Can you please recommend me some GSM gateway? I'm currently looking >>>>> > for a good one to buy... anyone have experience PORTech GSM gateways? >>>>> > Are they good? >>>>> > >>>>> > I also need it to work with FS, I'm also kinda new with VoIP >>>>> hardware. >>>>> > >>>>> > Thanks, >>>>> > >>>>> > Diego >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/f0d9a6aa/attachment-0001.html From darklion11 at yahoo.com Thu Jun 18 01:31:22 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 18 Jun 2009 01:31:22 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: References: <24045890.post@talk.nabble.com> <24065638.post@talk.nabble.com> Message-ID: <24088636.post@talk.nabble.com> Ok thanks a lot for that. Sorry my mistake.. Darren Schreiber wrote: > > Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn > of > "tcapi" still in the config file. > > If your test was: > # isql zenoss edmar edmar > > > Then zenoss should be your db_dsn: > > > Not > > > > You should be seeing something about the ODBC connection failing at > FreeSWITCH startup if you look at the log closely (search for > mod_nibblebill) that indicates this, too. > > - Darren > > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Tuesday, June 16, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to > ODBC > > > my nibble.conf.xml > > > > > > > param name="db_username" value="edmar"/> > param name="db_password" value="edmar"/> > param name="db_dsn" value="tcapi"/> > > > > > > param name="db_column_cash" value="cash"/> > > > param name="db_column_account" value="id"/> > > > > param name="global_heartbeat" value="1"/> > > !-- By default, warn a caller when their balance is at $5.00. You can > set this to a negative number. --> > param name="lowbal_amt" value="5"/> > param name="lowbal_action" value="play ding"/> > > > param name="nobal_amt" value="0"/> > param name="nobal_action" value="hangup"/> > > > param name="percall_max_amt" value="100"/> > param name="percall_action" value="hangup"/> > > > > > Account 1001.xml > > > > > param name="password" value="1234"/> > param name="vm-password" value="1001"/> > param name="vm-mailto" value=""/> > param name="vm-email-all-messages" value="false"/> > param name="vm-delete-file" value="false"/> > param name="vm-attach-file" value="false"/> > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > I check unixodbc has been installed. > > # isql zenoss edmar edmar > [SQL]> > > Connected successfully but on freeswitch error Cannot connect to user ODBC > [root] > > > Darren Schreiber wrote: >> >> What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the >> real logs from FS's logs? The info below is not nearly detailed enough. >> >> -----Original Message----- >> From: Edmar Cruz [mailto:darklion11 at yahoo.com] >> Sent: Monday, June 15, 2009 6:44 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to >> ODBC >> >> >> Hi >> >> I experiencing an error on mod_nibblebill. I already load it from >> autoload_configs, especially mod_spidermonkey. Uncomment >> mod_spidermonkey_odbc. I also download unixodbc and created the files >> /etc/odbcinst.ini and /etc/odbc.ini with the correct format >> >> [zenoss] >> DATABASE = tcapi >> USER = root >> PASS = password >> ..... >> >> I type also on the console isql zenoss root password. Also working... >> >> But an error occur on freeswitch Cannot connect to user [root] ... >> >> What do you thinks is the problem? >> -- >> View this message in context: >> > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 >> 890p24045890.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24065638.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24088636.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From panselva at gmail.com Thu Jun 18 03:06:44 2009 From: panselva at gmail.com (selva kumar) Date: Thu, 18 Jun 2009 15:36:44 +0530 Subject: [Freeswitch-users] Automatic call distribution Message-ID: <45f609f90906180306n2eec6ed3lb8cf84177faf2791@mail.gmail.com> Hi, I have setup FS for both inbound and outbound.It is working fine. Now I would like to configure Automatic Call Distribution(ACD).How to configure it in Freeswitch? Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/2c2964e7/attachment.html From steve at justfone.com Thu Jun 18 06:00:58 2009 From: steve at justfone.com (Steven Brown) Date: Thu, 18 Jun 2009 14:00:58 +0100 Subject: [Freeswitch-users] VAD, TALK and NOTALK events Message-ID: <3e6d7b0c0906180600q1b5f5b3dp4e28350de7aede98@mail.gmail.com> Hi, I have been trying to pick up TALK and NOTALK events but with no success, I have enabled VAD for "both" in my config and the rtp is stopping and starting as expected however when I hook up to the event socket and request "event talk notalk" nothing is ever fired, any thoughts on where I am going wrong appreciated. Thanks Steve Steven Brown email steve at justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. From brian at freeswitch.org Thu Jun 18 06:16:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 08:16:58 -0500 Subject: [Freeswitch-users] VAD, TALK and NOTALK events In-Reply-To: <3e6d7b0c0906180600q1b5f5b3dp4e28350de7aede98@mail.gmail.com> References: <3e6d7b0c0906180600q1b5f5b3dp4e28350de7aede98@mail.gmail.com> Message-ID: <866F2582-2968-4A33-9B1E-CCBBE6294FBA@freeswitch.org> I suspect you're going for TALK and NOTALK as the event names? its CUSTOM conference:: maintenance /b On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: > Hi, > > I have been trying to pick up TALK and NOTALK events but with no > success, I have enabled VAD for "both" in my config and the rtp is > stopping and starting as expected however when I hook up to the event > socket and request "event talk notalk" nothing is ever fired, any > thoughts on where I am going wrong appreciated. > > Thanks > > Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/e33c9a30/attachment.html From brian at freeswitch.org Thu Jun 18 06:44:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 08:44:56 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> Message-ID: <556D3D82-61EA-4ABB-ADEB-DF3B88B0F46B@freeswitch.org> If you're donating you can send it to my paypal brian at freeswitch.org, I also received the sound order for the zrtp sound files and a few odds and ends we needed. The order was 650 dollars and thus far I have only received a 50 dollar donation to help pay for it. So if you wanna pitch in on that also please let me know. I'm paying this out of my pocket. Thanks, Brian On Jun 17, 2009, at 1:56 PM, EdPimentl wrote: > I will match the 150.00 > > Best regards, > -E > CEO and Founder > Gpro.ws > http://Twitter.com/edpimentl > > http://TwebEX.com (Twitter Based Online Web Conference Platform) > http://TwitrShare.com (Send Picture and Message to Tweet Contacts) > http://TweetUp.ws (Twitter based MeetUp service) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/9a13f8d4/attachment.html From j3flight at gmail.com Thu Jun 18 06:55:07 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Thu, 18 Jun 2009 08:55:07 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <24084409.post@talk.nabble.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <24084409.post@talk.nabble.com> Message-ID: <4A3A473B.5090803@gmail.com> Wow, I apologize for the duplicate posts. The mailing list didn't want to cooperate with me last night... j3flight wrote: > I haven't gone to the trouble (yet) of making this work, but I believe you > could use execute_application from the conference controls to do just about > anything with JavaScript... > > Here's a wiki page I created after building a JavaScript IVR for a > conference server... > http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR > > There are a couple functions in there for voicing user count, etc. So, I > believe you could stick those in a script by themselves and call them from > execute_application. Somehow, you would have to identify what user is > calling the script and what conference they're in. (You could possibly set > a session variable upon entering the conference, or parse all the > conferences until you find that session's UUID.) > > I don't know what else you're trying to do, but once you get one of them > working, the rest should follow a similar template. > > Post back if you make it work, I'm interested! > From bjbrashier at gmail.com Thu Jun 18 07:26:55 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 07:26:55 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A39BD80.8020706@gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> Message-ID: <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> What I did last night was to go ahead and modify mod_conference.c to include a new "count" conference control. I've got it getting to the right place, and spitting debug messages with the right data about which member and what the count is, but for some reason the text-to-speech isn't working. That's what I'll be tacking today. The only other things I really need to figure out are a toggle for whether or not the moderator leaving ends the conference (from a DTMF, I have to clear all endconfs or something), and a command to mute all participants. Once I have those, I'm sure everything else will be a laydown. I'm not opposed to other methods, but I am opposed to increased complexity. If I can do it all in C and XML, I prefer that to some C, some XML, some lua, some JS, etc. I'll take a closer look at your example when I get into the office to see if that's a more elegant solution than what I have. On Wed, Jun 17, 2009 at 9:07 PM, wrote: > FYI: I fixed the Wiki documentation for the lock/unlock feature. > > Bradley Brashier wrote: > > So I found one interesting thing so far: the "lock" caller control > > actually does function as a toggle, and, in fact, "unlock" does not do > > anything. This goes against wiki docs on mod_conference, but is > > helpful in this instance. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/f430164a/attachment-0001.html From msc at freeswitch.org Thu Jun 18 07:31:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:31:10 -0500 Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <2557047.31431245309449504.JavaMail.root@zimbra.tng.de> References: <18011901.31411245309252614.JavaMail.root@zimbra.tng.de> <2557047.31431245309449504.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906180731h6f52b47cw72deaaafdc53370e@mail.gmail.com> Where is the pastebin with all of your configuration files? -MC On Thu, Jun 18, 2009 at 2:17 AM, Rudolf Denert wrote: > Yes, I removed the tags but with no effect. I think the problem is that the > webserver doesn't look in the directory where featuers.xml is deposited on > the freeswitchserver (/opt/freeswitch/conf/dialplan/). > > The issue is that FS finds the context when dialplan is the directory > /opt/freeswitch/conf/dialplan/public/ . > > But when it is on the webserver (it is on another server with a different > IP-address) I get the error that I told you. > > What should i verify in the default config. > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Brian West" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 > Amsterdam/Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Freeswitch / Webserver > > Its clearly telling you that context features doesn't exist... did you > remove the context tags around your extension so that it would be in > the correct context? Review the default config again. > > /b > > On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > > > Context features not found > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/da739caf/attachment.html From msc at freeswitch.org Thu Jun 18 07:33:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:33:29 -0500 Subject: [Freeswitch-users] Is freeswitch can call mobile phones? In-Reply-To: <24088222.post@talk.nabble.com> References: <24088222.post@talk.nabble.com> Message-ID: <87f2f3b90906180733w41698ec2pc8a5f0a3e5bc0619@mail.gmail.com> I am not aware of anyone who will give you free access to any kind of PSTN network. If you do find someone please let us in on the secret. :) -MC On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz wrote: > > Hi is there any possible free sites ip that i can connect so I can could to > any mobiles phones? > > I know some several ip sites has the capability to call for free Ip to > Voip... I know freeswitch can do this > > > Can you give me an example site? > -- > View this message in context: > http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/9de377b4/attachment.html From msc at freeswitch.org Thu Jun 18 07:35:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:35:21 -0500 Subject: [Freeswitch-users] Use Freeswitch to provide a SIP Video/Voice email service In-Reply-To: <24086865.post@talk.nabble.com> References: <24086865.post@talk.nabble.com> Message-ID: <87f2f3b90906180735w4616baa0k4899bbfcbbdcb04a@mail.gmail.com> On Thu, Jun 18, 2009 at 1:03 AM, Bilbo wrote: > > Hi, > > I would like to use Freeswitch to provide a Video/Voice mail service that > is > integrated with an email service. > > I would like to have the ability to email the Video/Voice messages as well > as the SIP users being able to collect their Video messages using their > video soft-phones. > > Has anyone done this before or know if Freeswitch is capable? > I'm sure that FS has all the hooks necessary, but it's like the proverbial Lego bricks: some assembly required. If someone has done this kind of thing already then we'd love to hear about it. -MC > > Thanks > -- > View this message in context: > http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-service-tp24086865p24086865.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/7de248f6/attachment.html From maxim.tsvetov at gmail.com Thu Jun 18 07:49:59 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Thu, 18 Jun 2009 18:49:59 +0400 Subject: [Freeswitch-users] CTI Message-ID: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> Hello! We are seeking possibilities to use CTI features with Freeswitch. This features are: - click-to-dial - call popup - answer call,hangup - call transfer Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..) or there is already written module or third-party software? This solution should support 100-150 simultaneous ?onnections from freeswitch users. Could you please share you experience with CTI. Regards, Maxim Tsvetov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/f0277967/attachment.html From msc at freeswitch.org Thu Jun 18 07:53:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:53:36 -0500 Subject: [Freeswitch-users] Automatic call distribution In-Reply-To: <45f609f90906180306n2eec6ed3lb8cf84177faf2791@mail.gmail.com> References: <45f609f90906180306n2eec6ed3lb8cf84177faf2791@mail.gmail.com> Message-ID: <87f2f3b90906180753g189fcc24s572fb32e160578d6@mail.gmail.com> On Thu, Jun 18, 2009 at 5:06 AM, selva kumar wrote: > Hi, > I have setup FS for both inbound and outbound.It is working fine. > Now I would like to configure Automatic Call Distribution(ACD).How to > configure it in Freeswitch? > > Start with this: http://wiki.freeswitch.org/wiki/Mod_fifo You can set up agents to be off-hook or on-hook and they can wait for calls. Enjoy! -MC > > Sam > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/6375b60a/attachment.html From msc at freeswitch.org Thu Jun 18 08:19:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 10:19:00 -0500 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! Message-ID: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/7f847a9b/attachment-0001.html From diego.viola at gmail.com Thu Jun 18 08:26:18 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 18 Jun 2009 11:26:18 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <698401620906180126n454589a7y3bc2ff4ae7004741@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> <698401620906180126n454589a7y3bc2ff4ae7004741@mail.gmail.com> Message-ID: <86a32abc0906180826y3c81599ese2279ba68810a3c@mail.gmail.com> Thanks for the suggestions guys, I think I will go with PORTech for now. @Jo?o Mesquita: Let me know when mod_khomp is done, I might consider getting some khomps in the future when the module is ready. Regards, Diego On Thu, Jun 18, 2009 at 4:26 AM, Jan Kubr wrote: > There are gateways that allow you to set your own caller ID? I thought it'd > always use the number of the SIM. > Jan > > On Thu, Jun 18, 2009 at 12:28 AM, jay binks wrote: >> >> Ive used these in the past. >> >> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >> >> sound fine, work well... >> reliable etc etc.. >> >> things to watch out for...? : >> >> *? cant send your own caller ID from them ( in my experience its locked to >> the sim ) >> *? your provider might block the IMEI number of the GSM terminal, if they >> dont like what your doing. >> >> just some stuff to consider. >> >> >> Jay >> >> >> >> >> 2009/6/18 Jo?o Mesquita >>> >>> Pricewise, is it worth it? >>> >>> jmesquita >>> >>> On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: >>>> >>>> We plan to buy one of these: >>>> >>>> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >>>> since you can use SMTP/POP3 to manage SMS. >>>> Jan >>>> 2009/6/17 Jo?o Mesquita >>>>> >>>>> Guys, I was looking at the advantages and disadvantages of having a GSM >>>>> gateway vs. a GSM board. >>>>> >>>>> The conclusions I get are: >>>>> >>>>> Board pros >>>>> >>>>> 1. Boards are able to get/send SMS without SIP "tricks" >>>>> 2. You don't have to make a SIP call to check if channel is available >>>>> and don't rely o SIP messages to get channel status >>>>> 3. FS will be able to check for signal level on the board and fire >>>>> events on pre-defined thresholds. >>>>> >>>>> Gateway pros >>>>> >>>>> 1. I think of is the a GW can be used by more then one server, >>>>> therefore, can have failover. >>>>> 2. A GW is more scalable >>>>> >>>>> It would be nice if you, that have already used GSM GWs in production, >>>>> could comment on this. >>>>> >>>>> Thanks, >>>>> >>>>> jmesquita >>>>> >>>>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> look at www.kuhnt.com. It?s a german page. There you can find >>>>>> "Kontakt" >>>>>> where you can ask for special requirements. >>>>>> >>>>>> NOx >>>>>> >>>>>> >>>>>> >>>>>> Diego Viola wrote: >>>>>> > >>>>>> > Hi everyone, >>>>>> > >>>>>> > Can you please recommend me some GSM gateway? I'm currently looking >>>>>> > for a good one to buy... anyone have experience PORTech GSM >>>>>> > gateways? >>>>>> > Are they good? >>>>>> > >>>>>> > I also need it to work with FS, I'm also kinda new with VoIP >>>>>> > hardware. >>>>>> > >>>>>> > Thanks, >>>>>> > >>>>>> > Diego >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Freeswitch-users mailing list >>>>>> > Freeswitch-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > >>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Sincerely >> >> Jay >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve at justfone.com Thu Jun 18 08:34:37 2009 From: steve at justfone.com (Steven Brown) Date: Thu, 18 Jun 2009 16:34:37 +0100 Subject: [Freeswitch-users] VAD, TALK and NOTALK events (Brian West) Message-ID: <3e6d7b0c0906180834r72dc14adre4e6b3ae25460576@mail.gmail.com> Thanks Brian, Yes I had been looking for TALK and NOTALK, CUSTOM conference::maintenance works great. Steve > > > Message: 4 > Date: Thu, 18 Jun 2009 08:16:58 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] VAD, TALK and NOTALK events > To: freeswitch-users at lists.freeswitch.org > Message-ID: <866F2582-2968-4A33-9B1E-CCBBE6294FBA at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > I suspect you're going for TALK and NOTALK as the event names? > > its CUSTOM conference:: maintenance > > /b > > > > On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: > > > Hi, > > > > I have been trying to pick up TALK and NOTALK events but with no > > success, I have enabled VAD for "both" in my config and the rtp is > > stopping and starting as expected however when I hook up to the event > > socket and request "event talk notalk" nothing is ever fired, any > > thoughts on where I am going wrong appreciated. > > > > Thanks > > > > Steve > > > > - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/a715869a/attachment.html From toofics at gmail.com Thu Jun 18 09:20:05 2009 From: toofics at gmail.com (Victor Toofic) Date: Thu, 18 Jun 2009 11:20:05 -0500 Subject: [Freeswitch-users] call quality problems in conference Message-ID: <1245342005.13905.39.camel@ktulu> Hi all! I'm having some troubles with call quality using conferences. The scenario is like this: An agent makes a call to freeswitch and enters in a conference room waiting for outbound calls; on the other side there is an application generating outbound calls and when one is answered it is assigned to the first agent available, so the outbound call enters in some of the agents's conference room (it is some kind of semi-predictive dialer). I'm using conferences because we need special features like monitoring or whispering to the agents. There are times when some of the outbound calls that enter in a conference room have really bad quality: broken/choppy voice, echo, etc, Something like this: http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav This occurs in 20%-40% of the outbound calls. I know it might be because of the jitter or packet loss with our voip provider. But.. this hardly occurs when the agents dial manually (using the bridge app); when dialing manually the problem (when it ocurrs) is always unperceptible. Thats why I think the conference room is aggravating the problem. Im using the 'jitterbuffer_msec=180' in the originate command and the same in the dialplan (when the agents log-in). What do you think is happening here? Am I missing something? Any guidance will be really appreciated! Thnks!! -- Regards.. Victor Toofic From brian at freeswitch.org Thu Jun 18 09:29:34 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 11:29:34 -0500 Subject: [Freeswitch-users] call quality problems in conference In-Reply-To: <1245342005.13905.39.camel@ktulu> References: <1245342005.13905.39.camel@ktulu> Message-ID: Please post bugs to http://jira.freeswitch.org /b On Jun 18, 2009, at 11:20 AM, Victor Toofic wrote: > Hi all! > > I'm having some troubles with call quality using conferences. The > scenario is like this: > > An agent makes a call to freeswitch and enters in a conference room > waiting for outbound calls; on the other side there is an application > generating outbound calls and when one is answered it is assigned to > the > first agent available, so the outbound call enters in some of the > agents's conference room (it is some kind of semi-predictive dialer). > > I'm using conferences because we need special features like monitoring > or whispering to the agents. > > There are times when some of the outbound calls that enter in a > conference room have really bad quality: broken/choppy voice, echo, > etc, > Something like this: > > http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav > > This occurs in 20%-40% of the outbound calls. I know it might be > because > of the jitter or packet loss with our voip provider. > > But.. this hardly occurs when the agents dial manually (using the > bridge > app); when dialing manually the problem (when it ocurrs) is always > unperceptible. Thats why I think the conference room is aggravating > the > problem. > > Im using the 'jitterbuffer_msec=180' in the originate command and the > same in the dialplan (when the agents log-in). > > What do you think is happening here? > Am I missing something? Any guidance will be really appreciated! > > Thnks!! > > -- > Regards.. > Victor Toofic > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saeedahmad1981 at gmail.com Thu Jun 18 09:40:32 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Thu, 18 Jun 2009 18:40:32 +0200 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! In-Reply-To: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> References: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> Message-ID: Done :) Guten Appetit On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins wrote: > Hello FreeSWITCHers out there! I have it on good authority that the > FreeSWITCH developers have all convened in an undisclosed location. Rumors > that they are plotting to take over the world are not yet confirmed but I > will keep you updated as information becomes available. :) > > It would be great for all of us to show our support and appreciation to the > guys for all the hard work they've done. How many of us have had a question > answered on the IRC channel or here on the list by one of the guys? How many > of us use FreeSWITCH every day for work? If you've benefited from their hard > work then please give a little. If we can get everyone to hop on the paypal > link (on http://www.freeswitch.org) and donate $5 or $10 then we can > easily pay for a nice dinner for the guys. > > Please hit the link and let me know (off list) when you've donated. Let's > do this, people! > > -Michael > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/0c9e5c85/attachment.html From msc at freeswitch.org Thu Jun 18 09:48:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 11:48:16 -0500 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! In-Reply-To: References: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> Message-ID: <87f2f3b90906180948i25031b67ga5668c4180b22820@mail.gmail.com> Thank you so much! The devs are really loving this. -MC On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad wrote: > Done :) > Guten Appetit > > On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins wrote: > >> Hello FreeSWITCHers out there! I have it on good authority that the >> FreeSWITCH developers have all convened in an undisclosed location. Rumors >> that they are plotting to take over the world are not yet confirmed but I >> will keep you updated as information becomes available. :) >> >> It would be great for all of us to show our support and appreciation to >> the guys for all the hard work they've done. How many of us have had a >> question answered on the IRC channel or here on the list by one of the guys? >> How many of us use FreeSWITCH every day for work? If you've benefited from >> their hard work then please give a little. If we can get everyone to hop on >> the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then >> we can easily pay for a nice dinner for the guys. >> >> Please hit the link and let me know (off list) when you've donated. Let's >> do this, people! >> >> -Michael >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/72e7641d/attachment-0001.html From andy at fabulous4.co.uk Thu Jun 18 09:54:30 2009 From: andy at fabulous4.co.uk (Andy) Date: Thu, 18 Jun 2009 17:54:30 +0100 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> Message-ID: <0DBFD71825C84229A9118D2D11A537D4@D810> Hi All, I've tested this new variable and everything works grand. I've tested recording to wav,mp3 and shoutcast and in all cases the sample rate is set correctly. I was about to post an entry on the wiki but I discovered a very similar variable already there called record_rate. I've tested this and it doesn't appear to work. Would you like me to replace this entry with details of the new one that does seem to work? A few more questions: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? Many thanks for sorting this one for me and for all your help. regards Andy _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 June 2009 18:52 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Fri, Jun 12, 2009 at 9:26 AM, Andy wrote: Excellent, thanks Anthony, I'll give it a go. Andy, can you report back on your success with this variable? Also, we would appreciate it if you could add an entry to the wiki on the channel_variables page. Let me know if you have any questions and I'll be glad to help. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/122e9c89/attachment.html From brian at freeswitch.org Thu Jun 18 10:10:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 12:10:35 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <0DBFD71825C84229A9118D2D11A537D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> <0DBFD71825C84229A9118D2D11A537D4@D810> Message-ID: <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> On Jun 18, 2009, at 11:54 AM, Andy wrote: > 1) I notice that when I change the sample rate it automatically > changes the bit rate too. I understand why this is the case but > wondered if it was just as easy to be able to control the bitrate as > well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. > 2) When I use a sample rate other than 8000 I get a warning 'Sample > rate doesn't match'. I guess this puts some extra load on the > server. If all my calls are being recorded and all at 11025 can/ > should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/697fdd1b/attachment.html From andy at fabulous4.co.uk Thu Jun 18 11:01:13 2009 From: andy at fabulous4.co.uk (Andy) Date: Thu, 18 Jun 2009 19:01:13 +0100 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> Message-ID: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5ec80288/attachment.html From mrene_lists at avgs.ca Thu Jun 18 11:08:05 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 18 Jun 2009 13:08:05 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). Math On 18-Jun-09, at 1:01 PM, Andy wrote: > Thanks Brian, > > So, just to calrify will the base call always be 8kHz? > > On a related note, do you happen to know the bitrate of each open > channel/live call? Is it 16 kilobits per second like the recorded > audio? I need to do some calculations on the badwidth required to > handle a certain number of concurrent calls. > > Many thanks > Andy > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: 18 June 2009 18:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Sample rate and recordFile > > > On Jun 18, 2009, at 11:54 AM, Andy wrote: > >> 1) I notice that when I change the sample rate it automatically >> changes the bit rate too. I understand why this is the case but >> wondered if it was just as easy to be able to control the bitrate >> as well as the sample rate. > > If you're talking about mod_shout, NO. You'll end up picking an > invalid bitrate and asking why it doesn't work... been there done > that... I changed it a few months back to pick the optimal bitrate > for the sample rate. > >> 2) When I use a sample rate other than 8000 I get a warning 'Sample >> rate doesn't match'. I guess this puts some extra load on the >> server. If all my calls are being recorded and all at 11025 can/ >> should I alter the sample rate of the base call to 11025? > > NO. Your phone call is running at 8kHz, Your sound file is 11025 > and they don't match, If you were to play this file into an 8k > channel without a resample it would sound a little like satan. or a > dragging tape deck. The file has to be resampled to match the > current session rate. > > /b > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/e2c5c56a/attachment-0001.html From brian at freeswitch.org Thu Jun 18 11:08:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 13:08:08 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: The call rates we support are 8, 16,32 and 48k /b On Jun 18, 2009, at 1:01 PM, Andy wrote: > Thanks Brian, > > So, just to calrify will the base call always be 8kHz? > > On a related note, do you happen to know the bitrate of each open > channel/live call? Is it 16 kilobits per second like the recorded > audio? I need to do some calculations on the badwidth required to > handle a certain number of concurrent calls. > > Many thanks > Andy > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5d2c9dcf/attachment.html From brian at freeswitch.org Thu Jun 18 11:08:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 13:08:43 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: <3EF08E69-1C64-44F9-B5AA-54BB1687A211@freeswitch.org> look in mod_shout you'll see my calculations.. I think it has to be multiples of 16 if I recall. /b On Jun 18, 2009, at 1:01 PM, Andy wrote: > > On a related note, do you happen to know the bitrate of each open > channel/live call? Is it 16 kilobits per second like the recorded > audio? I need to do some calculations on the badwidth required to > handle a certain number of concurrent calls. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/4ed51a0d/attachment.html From dave at 3c.co.uk Thu Jun 18 11:16:00 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 18 Jun 2009 21:16:00 +0300 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> <0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: <1245348960.21945.1.camel@dk-d820> - plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll have a bit of headroom available. --Dave > Most calls are at 8kHz. The formula for bandwidth is sampling rate * > bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). > > > Math > > On 18-Jun-09, at 1:01 PM, Andy wrote: > > > Thanks Brian, > > > > So, just to calrify will the base call always be 8kHz? > > > > On a related note, do you happen to know the bitrate of each open > > channel/live call? Is it 16 kilobits per second like the recorded > > audio? I need to do some calculations on the badwidth required to > > handle a certain number of concurrent calls. > > > > Many thanks > > Andy > > > > > > ____________________________________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Brian West > > Sent: 18 June 2009 18:11 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Sample rate and recordFile > > > > > > > > > > On Jun 18, 2009, at 11:54 AM, Andy wrote: > > > > > 1) I notice that when I change the sample rate it automatically > > > changes the bit rate too. I understand why this is the case but > > > wondered if it was just as easy to be able to control the bitrate > > > as well as the sample rate. > > > > > > If you're talking about mod_shout, NO. You'll end up picking an > > invalid bitrate and asking why it doesn't work... been there done > > that... I changed it a few months back to pick the optimal bitrate > > for the sample rate. > > > > > 2) When I use a sample rate other than 8000 I get a warning > > > 'Sample rate doesn't match'. I guess this puts some extra load on > > > the server. If all my calls are being recorded and all at 11025 > > > can/should I alter the sample rate of the base call to 11025? > > > > NO. Your phone call is running at 8kHz, Your sound file is 11025 > > and they don't match, If you were to play this file into an 8k > > channel without a resample it would sound a little like satan. or a > > dragging tape deck. The file has to be resampled to match the > > current session rate. > > > > > > /b > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From anthony.minessale at gmail.com Thu Jun 18 11:29:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 18 Jun 2009 13:29:10 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <1245348960.21945.1.camel@dk-d820> References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> <0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> <1245348960.21945.1.camel@dk-d820> Message-ID: <191c3a030906181129u5916ad85p5502dd80a8f7c604@mail.gmail.com> or go over the limit and you'll have Max Headroom =D On Thu, Jun 18, 2009 at 1:16 PM, David Knell wrote: > - plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll > have a bit of headroom available. > > --Dave > > > Most calls are at 8kHz. The formula for bandwidth is sampling rate * > > bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). > > > > > > Math > > > > On 18-Jun-09, at 1:01 PM, Andy wrote: > > > > > Thanks Brian, > > > > > > So, just to calrify will the base call always be 8kHz? > > > > > > On a related note, do you happen to know the bitrate of each open > > > channel/live call? Is it 16 kilobits per second like the recorded > > > audio? I need to do some calculations on the badwidth required to > > > handle a certain number of concurrent calls. > > > > > > Many thanks > > > Andy > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > > Brian West > > > Sent: 18 June 2009 18:11 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Sample rate and recordFile > > > > > > > > > > > > > > > On Jun 18, 2009, at 11:54 AM, Andy wrote: > > > > > > > 1) I notice that when I change the sample rate it automatically > > > > changes the bit rate too. I understand why this is the case but > > > > wondered if it was just as easy to be able to control the bitrate > > > > as well as the sample rate. > > > > > > > > > If you're talking about mod_shout, NO. You'll end up picking an > > > invalid bitrate and asking why it doesn't work... been there done > > > that... I changed it a few months back to pick the optimal bitrate > > > for the sample rate. > > > > > > > 2) When I use a sample rate other than 8000 I get a warning > > > > 'Sample rate doesn't match'. I guess this puts some extra load on > > > > the server. If all my calls are being recorded and all at 11025 > > > > can/should I alter the sample rate of the base call to 11025? > > > > > > NO. Your phone call is running at 8kHz, Your sound file is 11025 > > > and they don't match, If you were to play this file into an 8k > > > channel without a resample it would sound a little like satan. or a > > > dragging tape deck. The file has to be resampled to match the > > > current session rate. > > > > > > > > > /b > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/14c36ec4/attachment.html From nicolas at medularis.com Thu Jun 18 11:40:09 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 18 Jun 2009 14:40:09 -0400 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! In-Reply-To: <87f2f3b90906180948i25031b67ga5668c4180b22820@mail.gmail.com> References: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> <87f2f3b90906180948i25031b67ga5668c4180b22820@mail.gmail.com> Message-ID: <1b46b4e80906181140r5f83197bo57fccbd494232278@mail.gmail.com> Thank you for all the patience and effort. You've done a great work! Have a great meal! On Thu, Jun 18, 2009 at 12:48 PM, Michael Collins wrote: > Thank you so much! The devs are really loving this. > -MC > > > On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad wrote: > >> Done :) >> Guten Appetit >> >> On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins wrote: >> >>> Hello FreeSWITCHers out there! I have it on good authority that the >>> FreeSWITCH developers have all convened in an undisclosed location. Rumors >>> that they are plotting to take over the world are not yet confirmed but I >>> will keep you updated as information becomes available. :) >>> >>> It would be great for all of us to show our support and appreciation to >>> the guys for all the hard work they've done. How many of us have had a >>> question answered on the IRC channel or here on the list by one of the guys? >>> How many of us use FreeSWITCH every day for work? If you've benefited from >>> their hard work then please give a little. If we can get everyone to hop on >>> the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then >>> we can easily pay for a nice dinner for the guys. >>> >>> Please hit the link and let me know (off list) when you've donated. Let's >>> do this, people! >>> >>> -Michael >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/429a3e15/attachment-0001.html From mariouzae at gmail.com Thu Jun 18 09:58:38 2009 From: mariouzae at gmail.com (Mario Guerra Uzae da Silva -) Date: Thu, 18 Jun 2009 13:58:38 -0300 Subject: [Freeswitch-users] doubt of configuration Message-ID: Hi, I am new user of Freeswitch, I am having trouble doing basic configurations. Somebody could help me how to configure a simple extension? Thanks sorry for my bad english -- Mario Uzae -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/1980119a/attachment.html From diego.viola at gmail.com Thu Jun 18 12:21:33 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 18 Jun 2009 15:21:33 -0400 Subject: [Freeswitch-users] doubt of configuration In-Reply-To: References: Message-ID: <86a32abc0906181221t23ffc44aid8d2804aec7cd06f@mail.gmail.com> Sure, but you need to provide more details, what do you want to do exactly? On Thu, Jun 18, 2009 at 12:58 PM, Mario Guerra Uzae da Silva - < mariouzae at gmail.com> wrote: > Hi, I am new user of Freeswitch, I am having trouble doing basic > configurations. Somebody could help me how to configure a simple extension? > > Thanks > > sorry for my bad english > > -- > Mario Uzae > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/06f7cda2/attachment.html From nik.middleton at noblesolutions.co.uk Thu Jun 18 12:54:47 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 18 Jun 2009 20:54:47 +0100 Subject: [Freeswitch-users] high cpu utilization Message-ID: Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/77f89043/attachment.html From larclap at yahoo.com Thu Jun 18 12:54:53 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 18 Jun 2009 12:54:53 -0700 Subject: [Freeswitch-users] Some channel variables not written to cdr-csv? Message-ID: <016701c9f04e$a5f0b220$f1d21660$@com> I have defined the following template in autoload_config/cdr_csv.conf.xml: The resultant Master.csv in logs/1000.csv: "+19495551212","+19495551212","1000","default","2009-06-18 09:59:59","2009-06-18 10:00:06","2009-06-18 10:01:16","77","70","NORMAL_CLEARING","7551138e-5c29-11de-80e6-1b59605a543b" ,"75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+19 495551212 at 66.53.188.187","sofia/internal/sip:1001 at 192.168.10.101","" Both ${direction} and ${accountcode} do not have any data in the cdr file. Am I using the wrong variable names? I do see Caller-Direction with a valid value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx says that both these variables exist. Thanks for any help, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5671fee3/attachment.html From tntknight at gmail.com Thu Jun 18 13:30:40 2009 From: tntknight at gmail.com (Anthony Knight) Date: Thu, 18 Jun 2009 16:30:40 -0400 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: Message-ID: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Is this possibly an issue to do with a newer tickless kernel? see http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td23248559.html Tony On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > This one has me a little baffled. If have a recent build (in the last > week) of FS installed on two near identical HP servers. One happily runs > 400 concurrent calls at around 50% CPU. The other can only run around 50 > calls without the CPU going to 98%. Identical configs and lua script. > > > > Only diff is that the server having problems is running latest centos > 64bit, where the other is 32bit. Any suggestions of where I might start > looking? > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/46a6953a/attachment.html From kristian.kielhofner at gmail.com Thu Jun 18 13:36:11 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 18 Jun 2009 16:36:11 -0400 Subject: [Freeswitch-users] Bootable ISO Daily FreeSWITCH build Message-ID: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> Hello everyone, I've setup one of my build servers to do a fresh check out of SVN trunk and build AstLinux with it every day at 2AM EST. The ISO and build log (for the curious) are available here: http://mirror.astlinux.org/freeswitch/daily/ I just ran a test build but daily builds will begin showing up this evening/morning at 2AM. I plan on keeping about 30 days worth of ISO images. They should be bootable on just about anything including VMware and various other virtualization platforms. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From bjbrashier at gmail.com Thu Jun 18 14:23:10 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 14:23:10 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161458n42e9479are572b462387f8adb@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> <7bcfdd290906161458n42e9479are572b462387f8adb@mail.gmail.com> Message-ID: <7bcfdd290906181423xc474158h39268280cce20f5a@mail.gmail.com> So I rebooted, installed some OS updates, synched up, and am running again. I've also been doing closer comparisons between the conference I'm running and the same phones through VOIP to other locations (like between the phones without the conference). The lag isn't as bad as it was, a significant portion is due to the VOIP connection we've got (ie. conference aside), and yet certain phones still have more trouble through the conference than not, to the tune of at least 400ms more than the others. At this point, I'm prepared to punt -- blame the specific phones for now, and look at it again in a month or so when the project is closer to "done". But if anyone has any ideas on why certain phones would behave worse than others (a Polycom SoundPoint IP 320 SIP phone is the worst) I'm all ears. BB On Tue, Jun 16, 2009 at 2:58 PM, Bradley Brashier wrote: > Will do, just haven't had the time, yet! > > > On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> don't forget to read my suggestion too from earlier today =D >> >> >> >> On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon wrote: >> >>> I was able to reduce it considerably. I can?t say it is completely >>> gone but I am very confident the ~.5 second delay I hear is because of the >>> time it takes my voice to go through the leaps and bounds of the phone >>> company to our server. I had at least a 3-5 second delay before I >>> experimented with the conference settings. >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >>> Brashier >>> *Sent:* Tuesday, June 16, 2009 5:02 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* Re: [Freeswitch-users] Voice lag in conference >>> >>> >>> >>> I'm not sure I've got the opportunity to do that at the moment, but I do >>> appreciate the point of view of a fellow product user. Were you able to >>> eliminate noticeable lag, or just reduce it to reasonable levels? >>> >>> >>> >>> I'll try to do something similar when I update to the newest trunk as >>> Anthony suggested. My copy is only a week old, but I'll try whatever has a >>> chance of working, and I know you guys have been working on conferencing >>> (the Moderator function couldn't have been timed better for me!). >>> >>> On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon >>> wrote: >>> >>> I am not as knowledgeable as the developers that will respond to your >>> question but I had the same problem as you. Here is what I did to combat >>> the delay: >>> >>> >>> >>> First off I started everything from scratch. I reinstalled Linux and >>> then I reinstalled FreeSWITCH by creating .deb packages. >>> >>> I then created my own conference profile and set the sample rate to 4000 >>> and changed the energy level to 20. >>> >>> I also made sure to test the conference room from phones that were in >>> completely different areas so there wasn?t a chance for feedback or really >>> bad echoing problems. >>> >>> >>> >>> Once I knew the delay was solved I raised the sample rate to 8000. I >>> tested it to make sure it would work properly. >>> >>> >>> >>> As Michael stated, this could be your network infrastructure but I just >>> wanted to let another FreeSWITCH user know what I did to try and stop the >>> voice delay. >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >>> Brashier >>> *Sent:* Tuesday, June 16, 2009 1:52 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] Voice lag in conference >>> >>> >>> >>> I'm creating a conferencing product for use in a system with >>> theoretically several hundred concurrent calls. I'm using FreeSwitch to >>> create this product, but am not only new to FreeSwitch, but also the entire >>> telecom industry as well as Open Source projects in general (I'm a >>> recovering BIOS guy). >>> >>> I've got a bare-bones conference up and running on the server, including >>> a handshake and a couple of features, and am using the default packages from >>> the current trunk, but I've noticed that voice lag is a pretty big issue. >>> Common lag times are several hundred milliseconds, and I've heard as long as >>> a second. It seems to be at least marginally specific to individual phones >>> -- certain phones have longer lag than others even on the same call. >>> >>> My question is really about what my options are. Is this just a part of >>> SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim >>> down that will help? Is this a common issue? If it's common, is it expected >>> by the marketplace? >>> >>> This message contains confidential information and is intended only for >>> the individual named. If you are not the named addressee you should not >>> disseminate, distribute or copy this e-mail. Please notify the sender >>> immediately by e-mail if you have received this e-mail by mistake and delete >>> this e-mail from your system. E-mail transmission cannot be guaranteed to be >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, arrive late or incomplete, or contain viruses. The sender >>> therefore does not accept liability for any errors or omissions in the >>> contents of this message, which arise as a result of e-mail transmission. If >>> verification is required please request a hard-copy version. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> This message contains confidential information and is intended only for >>> the individual named. If you are not the named addressee you should not >>> disseminate, distribute or copy this e-mail. Please notify the sender >>> immediately by e-mail if you have received this e-mail by mistake and delete >>> this e-mail from your system. E-mail transmission cannot be guaranteed to be >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, arrive late or incomplete, or contain viruses. The sender >>> therefore does not accept liability for any errors or omissions in the >>> contents of this message, which arise as a result of e-mail transmission. If >>> verification is required please request a hard-copy version. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/b300fb3e/attachment-0001.html From bjbrashier at gmail.com Thu Jun 18 15:24:20 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 15:24:20 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> Message-ID: <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> OK, so I did some more experimenting today. I found a problem with the code I'm using (again, this is off the current trunk, but with some small modifications): conference_member_say in mod_conference.c is simply not working. There are several messages in there that can theoretically tell the user something, but all of them are bypassed in the vanilla build because the default profile plays a wav instead of generating them on the fly. If you take out the wav, the message is supposed to be generated. So I took out the wavs, but I'm not hearing any messages. BTW, something I discovered last week: straight out-of-the-box with no other modifications, if you make any changes to the set of "default" caller controls in conference.conf.xml, they don't get taken. The "default" caller controls appear to get overwritten in a hard-coded fashion in mod_conference.c. A "feature", perhaps, but very confusing for us new users. Can we add some documentation in there to that effect, perhaps? BB On Thu, Jun 18, 2009 at 7:26 AM, Bradley Brashier wrote: > What I did last night was to go ahead and modify mod_conference.c to > include a new "count" conference control. I've got it getting to the right > place, and spitting debug messages with the right data about which member > and what the count is, but for some reason the text-to-speech isn't working. > That's what I'll be tacking today. > > The only other things I really need to figure out are a toggle for whether > or not the moderator leaving ends the conference (from a DTMF, I have to > clear all endconfs or something), and a command to mute all participants. > Once I have those, I'm sure everything else will be a laydown. > > I'm not opposed to other methods, but I am opposed to increased complexity. > If I can do it all in C and XML, I prefer that to some C, some XML, some > lua, some JS, etc. I'll take a closer look at your example when I get into > the office to see if that's a more elegant solution than what I have. > > On Wed, Jun 17, 2009 at 9:07 PM, wrote: > >> FYI: I fixed the Wiki documentation for the lock/unlock feature. >> >> Bradley Brashier wrote: >> > So I found one interesting thing so far: the "lock" caller control >> > actually does function as a toggle, and, in fact, "unlock" does not do >> > anything. This goes against wiki docs on mod_conference, but is >> > helpful in this instance. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/73e471eb/attachment.html From evilla at chipoly.com Thu Jun 18 15:41:52 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Thu, 18 Jun 2009 16:41:52 -0600 Subject: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ? Message-ID: <00ca01c9f065$fa3af220$eeb0d660$@com> Hello, I am planning to build a plataform to sell content, pictures, tones, MMS, etc. Do you know wich GSM 3G boards should work? Anyone has done this? Greetings! Edwin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/50300801/attachment.html From d at unwire.it Thu Jun 18 16:26:29 2009 From: d at unwire.it (Darin Weeks) Date: Thu, 18 Jun 2009 16:26:29 -0700 Subject: [Freeswitch-users] Bootable ISO Daily FreeSWITCH build In-Reply-To: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> References: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> Message-ID: <989132e70906181626n117aea5fl84e987b337374d1a@mail.gmail.com> Thanks! I added a link from the wiki... http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux On Thu, Jun 18, 2009 at 1:36 PM, Kristian Kielhofner wrote: > Hello everyone, > > ?I've setup one of my build servers to do a fresh check out of SVN > trunk and build AstLinux with it every day at 2AM EST. ?The ISO and > build log (for the curious) are available here: > > http://mirror.astlinux.org/freeswitch/daily/ > > ?I just ran a test build but daily builds will begin showing up this > evening/morning at 2AM. > > ?I plan on keeping about 30 days worth of ISO images. ?They should be > bootable on just about anything including VMware and various other > virtualization platforms. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it From msc at freeswitch.org Thu Jun 18 16:35:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 18:35:46 -0500 Subject: [Freeswitch-users] Bootable ISO Daily FreeSWITCH build In-Reply-To: <989132e70906181626n117aea5fl84e987b337374d1a@mail.gmail.com> References: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> <989132e70906181626n117aea5fl84e987b337374d1a@mail.gmail.com> Message-ID: <87f2f3b90906181635l1c68c507q86574178c372f4a5@mail.gmail.com> Thanks for all of your help! On Thu, Jun 18, 2009 at 6:26 PM, Darin Weeks wrote: > Thanks! I added a link from the wiki... > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux > > On Thu, Jun 18, 2009 at 1:36 PM, Kristian > Kielhofner wrote: > > Hello everyone, > > > > I've setup one of my build servers to do a fresh check out of SVN > > trunk and build AstLinux with it every day at 2AM EST. The ISO and > > build log (for the curious) are available here: > > > > http://mirror.astlinux.org/freeswitch/daily/ > > > > I just ran a test build but daily builds will begin showing up this > > evening/morning at 2AM. > > > > I plan on keeping about 30 days worth of ISO images. They should be > > bootable on just about anything including VMware and various other > > virtualization platforms. > > > > -- > > Kristian Kielhofner > > http://www.astlinux.org > > http://blog.krisk.org > > http://www.star2star.com > > http://www.submityoursip.com > > http://www.voalte.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/a47b3b51/attachment.html From msc at freeswitch.org Thu Jun 18 16:38:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 18:38:19 -0500 Subject: [Freeswitch-users] Some channel variables not written to cdr-csv? In-Reply-To: <016701c9f04e$a5f0b220$f1d21660$@com> References: <016701c9f04e$a5f0b220$f1d21660$@com> Message-ID: <87f2f3b90906181638v47713ab0w490b8ba2f3f0fd00@mail.gmail.com> Do you have any way to ensure that those variables are populated? Can you manually set those in the dialplan? Also, are you doing a leg only or b leg only or both? -MC On Thu, Jun 18, 2009 at 2:54 PM, Lars Zeb wrote: > I have defined the following template in > autoload_config/cdr_csv.conf.xml: > > > > > > > > > > The resultant Master.csv in logs/1000.csv: > > > > "+19495551212","+19495551212","1000","default","2009-06-18 > 09:59:59","2009-06-18 10:00:06","2009-06-18 > 10:01:16","77","70","NORMAL_CLEARING","7551138e-5c29-11de-80e6-1b59605a543b","75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+ > 19495551212 at 66.53.188.187","sofia/internal/sip:1001 at 192.168.10.101 > ","" > > > > Both ${direction} and ${accountcode} do not have any data in the cdr file. > Am I using the wrong variable names? I do see Caller-Direction with a valid > value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki > at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx says > that both these variables exist. > > > > Thanks for any help, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/96feeb8f/attachment-0001.html From juanma.v82 at gmail.com Thu Jun 18 16:48:55 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Thu, 18 Jun 2009 20:48:55 -0300 Subject: [Freeswitch-users] Can it do it? Message-ID: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Hi, I need to have the hability to negotiate the codec in a session (using proxy media or bypass media), unfortunally I've been unable to achive this due the documentation that I've found about it's vague. I've already tried using "absolute_codec_string" and everything that says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams to ignore it when the media is in "bypass media" or "proxy media". I need to configure the FS as a SBC or as a pseudo proxy (I already know that FS is not intend to do it, but in the documentation says that it can). I've also tried to manually modify the SDP using: And updating variables "switch_r_sdp" and "switch_l_dsp" but it also seams to ignore it. Here is the config: Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 What I need, is to offer to the SWITCH only the codecs defined for Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, only offer the codecs available for Endpoint2. Eventually the SWITCH will do the transcoding. So, here is my question, is there any way to achive this? (Handle the invite codecs in bypass or proxy media), if so, is there any example to follow? o can you give a tip? Thanks in advance, Regards From j3flight at gmail.com Thu Jun 18 17:01:39 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Thu, 18 Jun 2009 19:01:39 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> Message-ID: <4A3AD563.50403@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/2e1b2651/attachment.html From bjbrashier at gmail.com Thu Jun 18 17:15:14 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 17:15:14 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A3AD563.50403@gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> Message-ID: <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> I was indeed looking at "announce-count", but from the code, it looks like that was designed to announce to the caller how many people were on the conference only when they were joining and the number was over a threshold specified in the profile. Not exactly what I was looking for, but it did help me find some of the right variables. And no, it didn't work, but I see now that it's most likely because conference_member_say wasn't working. I didn't think to try to define tts_engine and tts_voice, though, thinking that things like that had likely defaults. Obviously that would be an issue if not. I'll look at that next. Don't quote me on what announce-count is supposed to do, yet -- I only looked at it for long enough to tell that it wasn't what I needed. Once I have things working the way I want, I feel like I'll have enough data to be more certain of what everything does, and then I'll be happy to help you fill those out. I like your solution on the default controls. Naming them "sample" instead of "default" would do fine. Alternately, if we put a blurb in the comments above the default controls saying "these controls are hard-coded, and changes will not be taken into account. They are here as an example only", that would probably be good enough. Also, it's not clear that the DTMF commands for caller controls can be multiple digits. It might go without saying, but I didn't think about it until a little ways in, so something on the wiki might be nice. On Thu, Jun 18, 2009 at 5:01 PM, wrote: > I also saw the option for the "announce-count" conference parameter (which > i assume is what you're trying to use) and it didn't seem to work for me > either. I couldn't figure out whether I was doing something wrong or if it > was not working - that's why I implemented it in JS. Looking at the code > now, do you have tts_engine and tts_voice defined in the conference config > file. Looks like conference_member_say won't do anything without those... > > I can definitely attest to the confusion on your second point... It took > me a while to figure out the "default" conference controls as well. As long > as you name your caller-controls something else, it all works great. The > easy fix would be to modify the included conference config file so that the > sample conference controls had a different name. If someone removed them > manually, it would work as advertised. > > The wiki doc for mod_conference still needs some help too. I tried to fill > in what I knew recently by adding all the options I could find in the source > and re-arranging the page to make it easier to understand for new folks. I > had to leave a bunch of ??? in places though because I didn't know what > something did or meant... Can anyone help complete that? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/250cb78b/attachment.html From msc at freeswitch.org Thu Jun 18 17:51:14 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 18 Jun 2009 19:51:14 -0500 Subject: [Freeswitch-users] Last call: buy dinner for FreeSWITCH devs Message-ID: <59B37427-3A16-41F8-8696-EDECF4F2C9A3@freeswitch.org> FYI, the devs report that they are at the restaurant! Last chance to pitch in and feed the troops. :) hit the paypal button on the main FreeSWITCH page: http://www.freeswitch.org Keep those devs happy and fed and version 1.0.4 will be here before you know it! -MC From darklion11 at yahoo.com Thu Jun 18 18:25:58 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 18 Jun 2009 18:25:58 -0700 (PDT) Subject: [Freeswitch-users] Is freeswitch can call mobile phones? In-Reply-To: <87f2f3b90906180733w41698ec2pc8a5f0a3e5bc0619@mail.gmail.com> References: <24088222.post@talk.nabble.com> <87f2f3b90906180733w41698ec2pc8a5f0a3e5bc0619@mail.gmail.com> Message-ID: <24104115.post@talk.nabble.com> I got one... But its a secret... mercutioviz wrote: > > I am not aware of anyone who will give you free access to any kind of PSTN > network. If you do find someone please let us in on the secret. :) > > -MC > > On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz wrote: > >> >> Hi is there any possible free sites ip that i can connect so I can could >> to >> any mobiles phones? >> >> I know some several ip sites has the capability to call for free Ip to >> Voip... I know freeswitch can do this >> >> >> Can you give me an example site? >> -- >> View this message in context: >> http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24104115.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Jun 18 18:42:41 2009 From: dujinfang at gmail.com (seven) Date: Fri, 19 Jun 2009 09:42:41 +0800 Subject: [Freeswitch-users] Some channel variables not written to cdr-csv? In-Reply-To: <016701c9f04e$a5f0b220$f1d21660$@com> References: <016701c9f04e$a5f0b220$f1d21660$@com> Message-ID: try to run verbose_event before answer or bridge might help. On Jun 19, 2009, at 3:54 AM, Lars Zeb wrote: > I have defined the following template in autoload_config/ > cdr_csv.conf.xml: > > "${caller_id_name}","${caller_id_number}","$ > {destination_number}","${context}","${start_stamp}","${answer_stamp}", > "${end_stamp}","${duration}","${billsec}","${hangup_cause}","$ > {uuid}","${bleg_uuid}","${accountcode}","${read_codec}","$ > {write_codec}", > "${channel_name}","${bridge_channel}","${direction}" > > > The resultant Master.csv in logs/1000.csv: > > "+19495551212","+19495551212","1000","default","2009-06-18 > 09:59:59","2009-06-18 10:00:06","2009-06-18 > 10 > : > 01 > : > 16 > ","77 > ","70 > ","NORMAL_CLEARING > ","7551138e > -5c29 > -11de > -80e6 > -1b59605a543b > ","75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+19495551212 at 66.53.188.187 > ","sofia/internal/sip:1001 at 192.168.10.101","" > > Both ${direction} and ${accountcode} do not have any data in the cdr > file. Am I using the wrong variable names? I do see Caller-Direction > with a valid value ([inbound]) in freeswitch.log, but nothing like > accountcode. The wiki at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx > says that both these variables exist. > > Thanks for any help, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/616d0389/attachment-0001.html From timb0311 at hotmail.com Thu Jun 18 18:54:59 2009 From: timb0311 at hotmail.com (Tim B) Date: Thu, 18 Jun 2009 21:54:59 -0400 Subject: [Freeswitch-users] Originate fax to local extension for testing Message-ID: Trying to do a local test for faxing. Keep getting an error. Can someone tell me how to correct this? Tim default dialplan: //inbound from remote box works fine - connect asterisk box and fs box, then fax from asterisk to fs... OK - also fax from fs to asterisk.... OK // local fax on fs .... FAILS!! // my originate command: originate sofia/internal/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) // error as follows: 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing FreeSWITCH->8000 in context public 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 Legged calls 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] _________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/453d984d/attachment.html From paul.degt at gmail.com Thu Jun 18 19:06:05 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Thu, 18 Jun 2009 22:06:05 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available Message-ID: <4A3AF28D.4040708@gmail.com> http://versafon.com/versafonweb/Software.jsp Essentially it's a wrapper around inbound socket interface, not all events supported yet, and not all event parameters/variables. It's multi threaded and scaled well in testing. We offer commercial support and development for FreeSwitch as well. From jmesquita at gmail.com Thu Jun 18 19:30:56 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 18 Jun 2009 23:30:56 -0300 Subject: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ? In-Reply-To: <00ca01c9f065$fa3af220$eeb0d660$@com> References: <00ca01c9f065$fa3af220$eeb0d660$@com> Message-ID: <5a8712120906181930n50d66721m648159ee4a8cbc5a@mail.gmail.com> Right now, I am working on a board that will soon support all those features but it isn't compatible to FreeSWITCH just yet. Other then that, there was thread here before discussing PorTech GSM gateways. They might be able to help. If you are interested in using other platform with the Khomp boards, I can provide you a contact. Just get in touch with me offlist. Thanks, jmesquita On Thu, Jun 18, 2009 at 7:41 PM, Ing. Edwin Villarreal wrote: > Hello, I am planning to build a plataform to sell content, pictures, > tones, MMS, etc. > > > > Do you know wich GSM 3G boards should work? Anyone has done this? > > > > *Greetings!* > > *Edwin* > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/ae4bdf28/attachment.html From j3flight at gmail.com Thu Jun 18 19:38:44 2009 From: j3flight at gmail.com (j3flight) Date: Thu, 18 Jun 2009 19:38:44 -0700 (PDT) Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> Message-ID: <24104639.post@talk.nabble.com> As far as using multiple digits in the conference controls, that doesn't seem possible. I was hoping I could make all the commands require a preceding *, like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that could be added, but then you have other silly issues to worry about... i.e. what if someone defines *1 and *10? Anyway, the conference app is powerful, especially if you want to leverage the event socket (which I have yet to try, but I can tell that's where all the goodies are). Asterisk's MeetMe has more features out of the box, but is not nearly as easily customized. I feel like mod_conference needs the following things so new folks don't go cross-eyed trying to get it to work (and I'll be more than happy to assist with this where I can): -- if the TTS stuff is required for other features to work, it needs to be turned on by default (tts is built by default now, right?) -- a great number of the possible conference parameters are missing from the default config file. I've stuck all the possibilities on the wiki (with missing descriptions in many cases) but those need to be in the default config with better explanations. (or, it could be left off the wiki entirely and a link to the default config file could be used, so documentation is only kept in one place) -- Some explanation that the "default" caller controls are HARD-CODED. I'll take a look at the wiki in just a minute and clear it up, but the config file needs an explanation too. Maybe they should be commented (or removed entirely) just to prove that you get the default set of caller controls without them being defined...?? -- View this message in context: http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24104639.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Jun 18 20:01:23 2009 From: dujinfang at gmail.com (seven) Date: Fri, 19 Jun 2009 11:01:23 +0800 Subject: [Freeswitch-users] doubt of configuration In-Reply-To: References: Message-ID: <0C8D6930-D87D-40A1-9869-C0F47C59562A@gmail.com> extensions from 1000 - 1019 are available with password 1234 by default conf On Jun 19, 2009, at 12:58 AM, Mario Guerra Uzae da Silva - wrote: > Hi, I am new user of Freeswitch, I am having trouble doing basic > configurations. Somebody could help me how to configure a simple > extension? > > Thanks > > sorry for my bad english > > -- > Mario Uzae > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Jun 18 20:17:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 22:17:08 -0500 Subject: [Freeswitch-users] Last call: buy dinner for FreeSWITCH devs In-Reply-To: <59B37427-3A16-41F8-8696-EDECF4F2C9A3@freeswitch.org> References: <59B37427-3A16-41F8-8696-EDECF4F2C9A3@freeswitch.org> Message-ID: <48E5A4B0-5199-4366-937E-B56D8205F59E@freeswitch.org> I would like to thank everyone for Dinner... we had a great time... now MORE CODE!!! /b On Jun 18, 2009, at 7:51 PM, Michael S Collins wrote: > FYI, the devs report that they are at the restaurant! Last chance to > pitch in and feed the troops. :) hit the paypal button on the main > FreeSWITCH page: > http://www.freeswitch.org > > Keep those devs happy and fed and version 1.0.4 will be here before > you know it! > > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bjbrashier at gmail.com Thu Jun 18 20:30:00 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 20:30:00 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <24104639.post@talk.nabble.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> Message-ID: <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> I've been using multiple digits successfully right from the start, about 2 or 3 weeks ago. They do the separation of *1 and *10 the same way as several other systems -- by time. If you dial *, then 1, then wait past a timeout, then 0, you'll get *1, and *10 if you did it faster. I've tested by using 3 and 34 as separate commands, and I'm using * commands on my working system. Perhaps you should try again? Obviously, if you were confused, the docs on this could definitely be better. I'll check out the TTS stuff in the morning, and figure out those other parameters after that. Unless the original author wants to pipe up, of course. On Thu, Jun 18, 2009 at 7:38 PM, j3flight wrote: > > As far as using multiple digits in the conference controls, that doesn't > seem > possible. I was hoping I could make all the commands require a preceding > *, > like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that > could be added, but then you have other silly issues to worry about... > i.e. > what if someone defines *1 and *10? > > Anyway, the conference app is powerful, especially if you want to leverage > the event socket (which I have yet to try, but I can tell that's where all > the goodies are). Asterisk's MeetMe has more features out of the box, but > is not nearly as easily customized. > > I feel like mod_conference needs the following things so new folks don't go > cross-eyed trying to get it to work (and I'll be more than happy to assist > with this where I can): > -- if the TTS stuff is required for other features to work, it needs to be > turned on by default (tts is built by default now, right?) > -- a great number of the possible conference parameters are missing from > the > default config file. I've stuck all the possibilities on the wiki (with > missing descriptions in many cases) but those need to be in the default > config with better explanations. (or, it could be left off the wiki > entirely and a link to the default config file could be used, so > documentation is only kept in one place) > -- Some explanation that the "default" caller controls are HARD-CODED. > I'll > take a look at the wiki in just a minute and clear it up, but the config > file needs an explanation too. Maybe they should be commented (or removed > entirely) just to prove that you get the default set of caller controls > without them being defined...?? > -- > View this message in context: > http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24104639.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/cdbc021b/attachment-0001.html From msc at freeswitch.org Thu Jun 18 20:49:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 22:49:17 -0500 Subject: [Freeswitch-users] Originate fax to local extension for testing In-Reply-To: References: Message-ID: <87f2f3b90906182049v6fa04d89pcb8c8bc2fe305bc2@mail.gmail.com> Tim, We need some information, specifically we need you to turn on debugging at the console and give us the log from start of call to the very end. Go to the CLI and press F8 (or type "console loglevel debug") and then initiate the call. Capture everything from the CLI from start to finish, then drop it in a pb at pastebin.freeswitch.org. Send us back the pastebin number and we'll try and diagnose it. -MC On Thu, Jun 18, 2009 at 8:54 PM, Tim B wrote: > Trying to do a local test for faxing. Keep getting an error. Can someone > tell me how to correct this? > > Tim > > > > default dialplan: > > > > > > > > > > data="last_fax=${caller_id_number}-${strftime(%Y%m%d%H%M%S)}"/> > > > > > > > > > > > > //inbound from remote box works fine > > - connect asterisk box and fs box, then fax from asterisk to fs... OK > > - also fax from fs to asterisk.... OK > > > > // local fax on fs .... FAILS!! > > // my originate command: > > originate sofia/internal/8000 at 192.168.10.35&txfax(storage/fax/test.tif) > > > > // error as follows: > > 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing > FreeSWITCH->8000 in context public > 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 > Legged calls > 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 > at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > > > ------------------------------ > Insert movie times and more without leaving Hotmail?. See how. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/3d16dc66/attachment.html From dave at 3c.co.uk Thu Jun 18 21:31:35 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 19 Jun 2009 07:31:35 +0300 Subject: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ? In-Reply-To: <00ca01c9f065$fa3af220$eeb0d660$@com> References: <00ca01c9f065$fa3af220$eeb0d660$@com> Message-ID: <1245385895.4231.3.camel@dk-d820> Hi Edwin, Rather than using a GSM/3G card, you might do better to find a mobile services aggregator which covers the locations you're interested in - MBlox or Sybase365 would be two places to start - and use them. You'll get scalability, better reliability, etc.; be warned that MMS is *still* a pain. --Dave > Hello, I am planning to build a plataform to sell content, pictures, > tones, MMS, etc. > > > > Do you know wich GSM 3G boards should work? Anyone has done this? > > > > Greetings! > > Edwin > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From j3flight at gmail.com Thu Jun 18 21:45:41 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Thu, 18 Jun 2009 23:45:41 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> Message-ID: <4A3B17F5.9070400@gmail.com> Well crap, I must have had something else screwed up then with the multiple digits... I will try it out again soon, thanks for putting me back on track. I made some more changes to the wiki that hopefully clean up some confusion on a few things like the caller-controls. From bjbrashier at gmail.com Thu Jun 18 22:14:51 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 22:14:51 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A3B17F5.9070400@gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> Message-ID: <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> Actually, that's another good reason to do those wiki and/or code & comments changes... most likely, the reason you thought it couldn't be done is that you tried it and it didn't work... but you tried it on the default profile before you realized that it was hard coded. I know that's what I did and was confused about at first. I want to take a second and point out that while I may be complaining about some difficulties I'm having, the process has actually been FAR easier and faster than I had ever expected. This is a nice, solid product that works amazingly well amazingly quickly. I've been working with it for exactly 5 weeks now, starting from not knowing what SIP was or even that it had anything to do with VoIP. I've got a decent, demo conference bridge working, and am likely to be saving my company a good chunk of change as soon as I work out a few more kinks. A 2-month time to market from complete zero is just incredible. On Thu, Jun 18, 2009 at 9:45 PM, wrote: > Well crap, I must have had something else screwed up then with the > multiple digits... I will try it out again soon, thanks for putting me > back on track. I made some more changes to the wiki that hopefully > clean up some confusion on a few things like the caller-controls. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/05333b4a/attachment.html From mattdfong at gmail.com Thu Jun 18 23:16:35 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 18 Jun 2009 23:16:35 -0700 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | Message-ID: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> I have two providers and want to first try to originate the call with provider A, and if that fails on certain failure causes attempt to originate the same call with provider B. Normally I would do this using an | in the dial string like originate sofia/gatewayA/123456|sofia/gatewayB/123456 but I do not want it to fail over on failure codes like USER_BUSY or NO_ANSWER because then I'm simply wasting the second carrier's resources. instead I would like to set a which error codes are considered a failure. The wiki notes a failure_causes channel variable for bridged calls, but this does not seem to work in an originate statement like originate {failure_causes='RECOVERY_ON_TIMER_EXPIRE',continue_on_fail=false}sofia/gateway/ gatewaya.com/1XXXXXX |sofia/gateway/gatewayb.com/1XXXXXX 5000 Can anyone recommend a way to accomplish what I'm trying to do...preferably w/o mod_lcr? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/20a18063/attachment.html From mrene_lists at avgs.ca Thu Jun 18 23:19:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 19 Jun 2009 01:19:43 -0500 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> Message-ID: and then originate it. Math On 19-Jun-09, at 1:16 AM, Matthew Fong wrote: > I have two providers and want to first try to originate the call > with provider A, and if that fails on certain failure causes attempt > to originate the same call with provider B. > > Normally I would do this using an | in the dial string like > originate sofia/gatewayA/123456|sofia/gatewayB/123456 > > but I do not want it to fail over on failure codes like USER_BUSY or > NO_ANSWER because then I'm simply wasting the second carrier's > resources. instead I would like to set a which error codes are > considered a failure. The wiki notes a failure_causes channel > variable for bridged calls, but this does not seem to work in an > originate statement like > > originate > {failure_causes > ='RECOVERY_ON_TIMER_EXPIRE',continue_on_fail=false}sofia/gateway/ > gatewaya.com/1XXXXXX |sofia/gateway/gatewayb.com/1XXXXXX 5000 > > Can anyone recommend a way to accomplish what I'm trying to > do...preferably w/o mod_lcr? > > Thanks. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/733fe4bd/attachment.html From jason at jasonjgw.net Thu Jun 18 23:31:09 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 19 Jun 2009 16:31:09 +1000 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> Message-ID: <20090619063109.GA21842@jdc.jasonjgw.net> Mathieu Rene wrote: > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then > originate it. Or if you're originating from a script, set that as a channel variable first. From mattdfong at gmail.com Thu Jun 18 23:38:58 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 18 Jun 2009 23:38:58 -0700 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <20090619063109.GA21842@jdc.jasonjgw.net> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> <20090619063109.GA21842@jdc.jasonjgw.net> Message-ID: <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> the script is not part of a session or dial plan. :( On Thu, Jun 18, 2009 at 11:31 PM, Jason White wrote: > Mathieu Rene wrote: > > > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then > > originate it. > > Or if you're originating from a script, set that as a channel variable > first. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/caa4443b/attachment.html From jason at jasonjgw.net Thu Jun 18 23:43:25 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 19 Jun 2009 16:43:25 +1000 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> Message-ID: <20090619064325.GA22568@jdc.jasonjgw.net> Bradley Brashier wrote: > I want to take a second and point out that while I may be complaining about > some difficulties I'm having, the process has actually been FAR easier and > faster than I had ever expected. This is a nice, solid product that works > amazingly well amazingly quickly. I've been working with it for exactly 5 > weeks now, starting from not knowing what SIP was or even that it had > anything to do with VoIP. I've got a decent, demo conference bridge working, > and am likely to be saving my company a good chunk of change as soon as I > work out a few more kinks. A 2-month time to market from complete zero is > just incredible. If organizations such as yours could contribute some of the money they save to FreeSWITCH development, for example to get desired features implemented or bugs fixed, I'm sure that would help to make what is already a great project even better. The software is excellent, the community helpful, and I've had a lot of fun experimenting with IPv6, TLS, ZRTP and other features that few other projects have implemented. From nik.middleton at noblesolutions.co.uk Fri Jun 19 01:23:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 19 Jun 2009 09:23:20 +0100 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I don't think that's the issue Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Knight Sent: 18 June 2009 21:31 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] high cpu utilization Is this possibly an issue to do with a newer tickless kernel? see http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td2324855 9.html Tony On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton wrote: Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/be51d9a6/attachment.html From timb0311 at hotmail.com Fri Jun 19 02:31:08 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 05:31:08 -0400 Subject: [Freeswitch-users] Originate fax to local extension for testing In-Reply-To: References: Message-ID: Michael, I ran the debugging you asked. I also tried to post it to pastebin.freeswitch.org but can't login. I used my login for the freeswitch site, but that doesn't seem to work?? How do I gain acess? Thanks. Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 18 Jun 2009 22:49:17 -0500 > From: Michael Collins > Subject: Re: [Freeswitch-users] Originate fax to local extension for > testing > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906182049v6fa04d89pcb8c8bc2fe305bc2 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Tim, > > We need some information, specifically we need you to turn on debugging at > the console and give us the log from start of call to the very end. Go to > the CLI and press F8 (or type "console loglevel debug") and then initiate > the call. Capture everything from the CLI from start to finish, then drop it > in a pb at pastebin.freeswitch.org. Send us back the pastebin number and > we'll try and diagnose it. > > -MC > _________________________________________________________________ Hotmail? has ever-growing storage! Don?t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/df43d281/attachment.html From jason at jasonjgw.net Fri Jun 19 02:43:46 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 19 Jun 2009 19:43:46 +1000 Subject: [Freeswitch-users] Originate fax to local extension for testing In-Reply-To: References: Message-ID: <20090619094346.GA8755@jdc.jasonjgw.net> Tim B wrote: > > Michael, I ran the debugging you asked. I also tried to post it to > pastebin.freeswitch.org but can't login. I used my login for the freeswitch > site, but that doesn't seem to work?? How do I gain acess? When I connect to pastebin.freeswitch.org I get a helpful notice saying the login and password is pastebin/freeswitch From darklion11 at yahoo.com Fri Jun 19 04:12:55 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 19 Jun 2009 04:12:55 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? Message-ID: <24109532.post@talk.nabble.com> My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/1006 at 116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1006 at 116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1006 at 116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 I put this on my external and internal profile And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24109532.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Fri Jun 19 04:13:30 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 19 Jun 2009 04:13:30 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? Message-ID: <24109532.post@talk.nabble.com> My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/1006 at 116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1006 at 116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1006 at 116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 I put this on my external and internal profile param name="apply-inbound-acl" value="globals"/> And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24109532.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From saeedahmad1981 at gmail.com Fri Jun 19 05:06:46 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 19 Jun 2009 14:06:46 +0200 Subject: [Freeswitch-users] Can it do it? In-Reply-To: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Message-ID: It seems that you want to transcode G729 which is not possible. You can use it in passthru mode only. On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote: > Hi, > > I need to have the hability to negotiate the codec in a session (using > proxy media or bypass media), unfortunally I've been unable to achive > this due the documentation that I've found about it's vague. > > I've already tried using "absolute_codec_string" and everything that > says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams > to ignore it when the media is in "bypass media" or "proxy media". I > need to configure the FS as a SBC or as a pseudo proxy (I already know > that FS is not intend to do it, but in the documentation says that it > can). > > I've also tried to manually modify the SDP using: > > > And updating variables "switch_r_sdp" and "switch_l_dsp" but it also > seams to ignore it. > > Here is the config: > > Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 > > What I need, is to offer to the SWITCH only the codecs defined for > Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, > only offer the codecs available for Endpoint2. Eventually the SWITCH > will do the transcoding. > > So, here is my question, is there any way to achive this? (Handle the > invite codecs in bypass or proxy media), if so, is there any example > to follow? o can you give a tip? > > Thanks in advance, > Regards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/480ba462/attachment.html From saeedahmad1981 at gmail.com Fri Jun 19 05:08:50 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 19 Jun 2009 14:08:50 +0200 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Message-ID: BTW, what is SWITCH? Can it do transcoding? On Fri, Jun 19, 2009 at 2:06 PM, Saeed Ahmad wrote: > It seems that you want to transcode G729 which is not possible. You can use > it in passthru mode only. > > > On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote: > >> Hi, >> >> I need to have the hability to negotiate the codec in a session (using >> proxy media or bypass media), unfortunally I've been unable to achive >> this due the documentation that I've found about it's vague. >> >> I've already tried using "absolute_codec_string" and everything that >> says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams >> to ignore it when the media is in "bypass media" or "proxy media". I >> need to configure the FS as a SBC or as a pseudo proxy (I already know >> that FS is not intend to do it, but in the documentation says that it >> can). >> >> I've also tried to manually modify the SDP using: >> >> >> And updating variables "switch_r_sdp" and "switch_l_dsp" but it also >> seams to ignore it. >> >> Here is the config: >> >> Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 >> >> What I need, is to offer to the SWITCH only the codecs defined for >> Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, >> only offer the codecs available for Endpoint2. Eventually the SWITCH >> will do the transcoding. >> >> So, here is my question, is there any way to achive this? (Handle the >> invite codecs in bypass or proxy media), if so, is there any example >> to follow? o can you give a tip? >> >> Thanks in advance, >> Regards >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/fa5d19cb/attachment.html From timb0311 at hotmail.com Fri Jun 19 05:39:13 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 08:39:13 -0400 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 In-Reply-To: References: Message-ID: here is the log... http://pastebin.freeswitch.org/9440 haha, yeah i see it now... duh. pulled an all nighter, too many things going on. must have overlooked it. > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > the login and password is pastebin/freeswitch > been trying to break myself into freeswitch on top of my original workload. thanks for the help. _________________________________________________________________ Lauren found her dream laptop. Find the PC that?s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c53ba6fc/attachment.html From brian at freeswitch.org Fri Jun 19 06:06:31 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Jun 2009 08:06:31 -0500 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> <20090619063109.GA21842@jdc.jasonjgw.net> <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> Message-ID: <58F248FD-6963-4171-9589-92D506DC6935@freeswitch.org> If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT issue. /b On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote: > the script is not part of a session or dial plan. :( > > On Thu, Jun 18, 2009 at 11:31 PM, Jason White > wrote: > Mathieu Rene wrote: > > > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then > > originate it. > > Or if you're originating from a script, set that as a channel > variable first. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/1f7ccb7c/attachment.html From anthony.minessale at gmail.com Fri Jun 19 07:51:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Jun 2009 09:51:35 -0500 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: <191c3a030906190751m9e04eb4lcff5463534863372@mail.gmail.com> use top -H to get the per-thread cpu usage. see if any one thread is using more than the rest. then get a gcore of the running process and do a thread apply all bt and get a bt from the thread with the matching id. Maybe that will tell you what is doing all the work. On Fri, Jun 19, 2009 at 3:23 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I?m running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly > I don?t think that?s the issue > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Knight > *Sent:* 18 June 2009 21:31 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] high cpu utilization > > > > Is this possibly an issue to do with a newer tickless kernel? > > > > see > http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td23248559.html > > > > Tony > > On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi Guys, > > > > This one has me a little baffled. If have a recent build (in the last > week) of FS installed on two near identical HP servers. One happily runs > 400 concurrent calls at around 50% CPU. The other can only run around 50 > calls without the CPU going to 98%. Identical configs and lua script. > > > > Only diff is that the server having problems is running latest centos > 64bit, where the other is 32bit. Any suggestions of where I might start > looking? > > > > Regards, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/3586f84c/attachment-0001.html From msc at freeswitch.org Fri Jun 19 08:00:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 10:00:35 -0500 Subject: [Freeswitch-users] Update - Transmit fax locally for test Message-ID: <87f2f3b90906190800u5d9436cbu2bd594bc8d09503@mail.gmail.com> Tim, Look at lines 47 and 48 of the pastebin. I think something goofy is happening there. What is 8000 at x.x.x.x in your system? Is that the receive fax extension? -MC ---------- Forwarded message ---------- From: Tim B Date: Fri, Jun 19, 2009 at 7:39 AM Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 To: freeswitch-users at lists.freeswitch.org here is the log... http://pastebin.freeswitch.org/9440 haha, yeah i see it now... duh. pulled an all nighter, too many things going on. must have overlooked it. > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > the login and password is pastebin/freeswitch > been trying to break myself into freeswitch on top of my original workload. thanks for the help. ------------------------------ Lauren found her dream laptop. Find the PC that?s right for you. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c9567fad/attachment.html From max.bridgewater at gmail.com Fri Jun 19 08:14:58 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 11:14:58 -0400 Subject: [Freeswitch-users] Help with Socket event again Message-ID: Any help our there? I'm still trying to get this piece working. Essentially what i wan to do is, when a call comes in (from registered devices as well as unregistered devices), notify the my server socket. Somehow it's not working. The change i made compared to the standard Freeswitch settings are the following: 1) Added following extension that in /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: 2) Changed file: /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: I noticed that with this extension, all calls received from external providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. But calls from registered devices and initiated using the socket interface are not forwarded. Is there something that need to be changed in the profiles? or is something wrong with my dial string? {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. In the logs, i cannot see that that my extension is being matched. Any idea, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/b1ac05b3/attachment.html From apt.get at gmail.com Fri Jun 19 08:19:59 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 19 Jun 2009 09:19:59 -0600 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: On Fri, Jun 19, 2009 at 2:23 AM, Nik Middleton wrote: > I?m running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I > don?t think that?s the issue I could be wrong, but I think PAE is a 32-bit kernel adapted for hardware with >4GB RAM. This can create a lot of overhead compared to running a true 64-bit kernel or a 32-bit kernel without PAE. Confirm this with 'uname -a' db From msc at freeswitch.org Fri Jun 19 08:43:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 10:43:10 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: Message-ID: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> Can you turn on debugging (F8) and capture all the output after your originate? Put it into a pastebin. (pastebin.freeswitch.org) -MC On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater wrote: > Any help our there? > > I'm still trying to get this piece working. Essentially what i wan to do > is, when a call comes in (from registered devices as well as unregistered > devices), notify the my server socket. Somehow it's not working. The change > i made compared to the standard Freeswitch settings are the following: > > 1) Added following extension that in > /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: > > > > > > /> > > > > > 2) Changed file: > /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: > > > > > > > > > > > > > I noticed that with this extension, all calls received from external > providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. > But calls from registered devices and initiated using the socket interface > are not forwarded. Is there something that need to be changed in the > profiles? > > or is something wrong with my dial string? > {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. > > In the logs, i cannot see that that my extension is being matched. > > Any idea, > > Max. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/28c973c5/attachment.html From bjbrashier at gmail.com Fri Jun 19 08:59:57 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 08:59:57 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <20090619064325.GA22568@jdc.jasonjgw.net> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> Message-ID: <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I chose flite), uncommenting a few things, and setting the TTS variables in the conference profile. You do have to rebuild FS to do this. With that working, my count function works, too. I posted a bug last night about conferencing, BTW: if you're using the wait-mod function, where the conference doesn't start until the moderator arrives, and you're also using separate profiles for users and moderators, the users only have the "user" profile until the first moderator arrives. At that time, they switch to also be using the "moderator" profile. Now that TTS is working, I'm going to see about helping you fill out those ???s, and maybe see if I can figure out how to fix the above bug. BB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c38c19e6/attachment.html From mattdfong at gmail.com Fri Jun 19 10:35:37 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 10:35:37 -0700 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <58F248FD-6963-4171-9589-92D506DC6935@freeswitch.org> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> <20090619063109.GA21842@jdc.jasonjgw.net> <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> <58F248FD-6963-4171-9589-92D506DC6935@freeswitch.org> Message-ID: <4256bf830906191035x7df2b635q699d503dc96b615f@mail.gmail.com> recovery_on_timer_expire was just my example.. I actually just want to try carrier B on everything except no_answer or user_busy... On Fri, Jun 19, 2009 at 6:06 AM, Brian West wrote: > If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT issue. > /b > > On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote: > > the script is not part of a session or dial plan. :( > > On Thu, Jun 18, 2009 at 11:31 PM, Jason White wrote: > >> Mathieu Rene wrote: >> > > > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then >> > originate it. >> >> Or if you're originating from a script, set that as a channel variable >> first. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/d81b1108/attachment-0001.html From max.bridgewater at gmail.com Fri Jun 19 10:58:59 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 13:58:59 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> Message-ID: Hi Mike, It's pasted here: http://pastebin.ca/1466521 Thanks, Max. On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: > Can you turn on debugging (F8) and capture all the output after your > originate? Put it into a pastebin. (pastebin.freeswitch.org) > -MC > > On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Any help our there? >> >> I'm still trying to get this piece working. Essentially what i wan to do >> is, when a call comes in (from registered devices as well as unregistered >> devices), notify the my server socket. Somehow it's not working. The change >> i made compared to the standard Freeswitch settings are the following: >> >> 1) Added following extension that in >> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >> >> >> >> >> >> > /> >> >> >> >> >> 2) Changed file: >> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >> >> >> >> >> >> >> >> >> >> >> >> >> I noticed that with this extension, all calls received from external >> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >> But calls from registered devices and initiated using the socket interface >> are not forwarded. Is there something that need to be changed in the >> profiles? >> >> or is something wrong with my dial string? >> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >> >> In the logs, i cannot see that that my extension is being matched. >> >> Any idea, >> >> Max. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/4d8f755d/attachment.html From msc at freeswitch.org Fri Jun 19 11:10:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 13:10:56 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> Message-ID: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> Max, that pastebin failed miserably as none of the xml shows up. can you try again or use our pastebin.freeswitch.org site? -MC On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater wrote: > Hi Mike, > > It's pasted here: http://pastebin.ca/1466521 > > Thanks, > Max. > > > > > On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: > >> Can you turn on debugging (F8) and capture all the output after your >> originate? Put it into a pastebin. (pastebin.freeswitch.org) >> -MC >> >> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Any help our there? >>> >>> I'm still trying to get this piece working. Essentially what i wan to do >>> is, when a call comes in (from registered devices as well as unregistered >>> devices), notify the my server socket. Somehow it's not working. The change >>> i made compared to the standard Freeswitch settings are the following: >>> >>> 1) Added following extension that in >>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>> >>> >>> >>> >>> >>> >> /> >>> >>> >>> >>> >>> 2) Changed file: >>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I noticed that with this extension, all calls received from external >>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>> But calls from registered devices and initiated using the socket interface >>> are not forwarded. Is there something that need to be changed in the >>> profiles? >>> >>> or is something wrong with my dial string? >>> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >>> >>> In the logs, i cannot see that that my extension is being matched. >>> >>> Any idea, >>> >>> Max. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/49b53531/attachment.html From max.bridgewater at gmail.com Fri Jun 19 11:19:09 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 14:19:09 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> Message-ID: Mike, Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to me though. Max. On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: > Max, > that pastebin failed miserably as none of the xml shows up. can you try > again or use our pastebin.freeswitch.org site? > -MC > > > On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Hi Mike, >> >> It's pasted here: http://pastebin.ca/1466521 >> >> Thanks, >> Max. >> >> >> >> >> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >> >>> Can you turn on debugging (F8) and capture all the output after your >>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>> -MC >>> >>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>> max.bridgewater at gmail.com> wrote: >>> >>>> Any help our there? >>>> >>>> I'm still trying to get this piece working. Essentially what i wan to do >>>> is, when a call comes in (from registered devices as well as unregistered >>>> devices), notify the my server socket. Somehow it's not working. The change >>>> i made compared to the standard Freeswitch settings are the following: >>>> >>>> 1) Added following extension that in >>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2) Changed file: >>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I noticed that with this extension, all calls received from external >>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>> But calls from registered devices and initiated using the socket interface >>>> are not forwarded. Is there something that need to be changed in the >>>> profiles? >>>> >>>> or is something wrong with my dial string? >>>> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >>>> >>>> >>>> In the logs, i cannot see that that my extension is being matched. >>>> >>>> Any idea, >>>> >>>> Max. >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/f3d8db96/attachment-0001.html From bjbrashier at gmail.com Fri Jun 19 11:27:00 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 11:27:00 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> Message-ID: <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> So it turns out that it wasn't a bug at all -- it is a feature that was not implemented. So I've got some work to do to get that running. Since I said I would, though, here's my analysis of the conference parameters you were asking about: mute-detect-sound Different sound for if muting using only when mute-detect flag is on. max-members Specifies the maximum number of participants in a call. max-members-sound If caller cannot join because max is reached, this sound plays. Recommended if max-members is set. comfort-noise-level Sets volume of background white noise to generate. announce-count Requires TTS. When joining, tells caller how many callers are already in conference if at least the specified minimum. suppress-events ? Sets a flag, but does not appear to do anything with it. verbose-events Maximum verbosity for transcripting. timer-name Specifies the name of this profile's timer. To separate it from other timers? BB On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier wrote: > OK, I figured out the TTS stuff. It's a matter of choosing an engine (I > chose flite), uncommenting a few things, and setting the TTS variables in > the conference profile. You do have to rebuild FS to do this. > > With that working, my count function works, too. > > I posted a bug last night about conferencing, BTW: if you're using the > wait-mod function, where the conference doesn't start until the moderator > arrives, and you're also using separate profiles for users and moderators, > the users only have the "user" profile until the first moderator arrives. At > that time, they switch to also be using the "moderator" profile. > > Now that TTS is working, I'm going to see about helping you fill out those > ???s, and maybe see if I can figure out how to fix the above bug. > > BB > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/4ba96921/attachment.html From bjbrashier at gmail.com Fri Jun 19 11:47:16 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 11:47:16 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> Message-ID: <7bcfdd290906191147n5742fc75mdab6525c67233500@mail.gmail.com> >mute-detect-sound >Different sound for if muting using only when mute-detect flag is on. Sorry... that didn't come out right. Try this: Different sound to play when muted by someone else and mute-detect flag is on. Plays mute-sound if this is not present. On Fri, Jun 19, 2009 at 11:27 AM, Bradley Brashier wrote: > So it turns out that it wasn't a bug at all -- it is a feature that was not > implemented. So I've got some work to do to get that running. Since I said I > would, though, here's my analysis of the conference parameters you were > asking about: > > mute-detect-sound > Different sound for if muting using only when mute-detect flag is on. > max-members > Specifies the maximum number of participants in a call. > max-members-sound > If caller cannot join because max is reached, this sound plays. Recommended > if max-members is set. > comfort-noise-level > Sets volume of background white noise to generate. > announce-count > Requires TTS. When joining, tells caller how many callers are already in > conference if at least the specified minimum. > suppress-events > ? Sets a flag, but does not appear to do anything with it. > verbose-events > Maximum verbosity for transcripting. > timer-name > Specifies the name of this profile's timer. To separate it from other > timers? > > BB > On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier wrote: > >> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I >> chose flite), uncommenting a few things, and setting the TTS variables in >> the conference profile. You do have to rebuild FS to do this. >> >> With that working, my count function works, too. >> >> I posted a bug last night about conferencing, BTW: if you're using the >> wait-mod function, where the conference doesn't start until the moderator >> arrives, and you're also using separate profiles for users and moderators, >> the users only have the "user" profile until the first moderator arrives. At >> that time, they switch to also be using the "moderator" profile. >> >> Now that TTS is working, I'm going to see about helping you fill out those >> ???s, and maybe see if I can figure out how to fix the above bug. >> >> BB >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/10c9b207/attachment.html From juanma.v82 at gmail.com Fri Jun 19 12:11:58 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Fri, 19 Jun 2009 16:11:58 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Message-ID: <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> Saeed Ahmad: Yes, it can do transcoding. Transcoding isn't the problem to my architecture, my problem is the codec negotiation between FS and Endpoints. I want to use FS as SBC (session border controller) or pseudo SIP proxy. How i said in my last e-mail FS will work in bypass-media or proxy-media in both modes FS doesn't handle media. I only want the FS handle Codec Negotiation how i described. Thanks On 19/06/2009, at 09:08, Saeed Ahmad wrote: > BTW, what is SWITCH? Can it do transcoding? > > On Fri, Jun 19, 2009 at 2:06 PM, Saeed Ahmad > wrote: > It seems that you want to transcode G729 which is not possible. You > can use it in passthru mode only. > > > On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote: > Hi, > > I need to have the hability to negotiate the codec in a session (using > proxy media or bypass media), unfortunally I've been unable to achive > this due the documentation that I've found about it's vague. > > I've already tried using "absolute_codec_string" and everything that > says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams > to ignore it when the media is in "bypass media" or "proxy media". I > need to configure the FS as a SBC or as a pseudo proxy (I already know > that FS is not intend to do it, but in the documentation says that it > can). > > I've also tried to manually modify the SDP using: > > > And updating variables "switch_r_sdp" and "switch_l_dsp" but it also > seams to ignore it. > > Here is the config: > > Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 > > What I need, is to offer to the SWITCH only the codecs defined for > Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, > only offer the codecs available for Endpoint2. Eventually the SWITCH > will do the transcoding. > > So, here is my question, is there any way to achive this? (Handle the > invite codecs in bypass or proxy media), if so, is there any example > to follow? o can you give a tip? > > Thanks in advance, > Regards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/3432d413/attachment.html From mattdfong at gmail.com Fri Jun 19 12:15:37 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 12:15:37 -0700 Subject: [Freeswitch-users] CHANNEL_HANGUP event no longer having variable_billsec in header Message-ID: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> I upgraded to 13857 today, but noticed that the channel_hangup event no longer contain the variable_billsec header. Is this correct, or am I crazy? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c7faec54/attachment.html From brian at freeswitch.org Fri Jun 19 12:19:46 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Jun 2009 14:19:46 -0500 Subject: [Freeswitch-users] Can it do it? In-Reply-To: <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> Message-ID: <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> No right now you can not legally transcode G729 in FreeSWITCH, PERIOD! /b On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > Yes, it can do transcoding. Transcoding isn't the problem to my > architecture, my problem is the codec negotiation between FS and > Endpoints. > From jmesquita at gmail.com Fri Jun 19 13:05:17 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 19 Jun 2009 17:05:17 -0300 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> Message-ID: <5a8712120906191305x5a98228eyb1847cfd2905aab5@mail.gmail.com> Inline... On Fri, Jun 19, 2009 at 3:27 PM, Bradley Brashier wrote: > So it turns out that it wasn't a bug at all -- it is a feature that was not > implemented. So I've got some work to do to get that running. Since I said I > would, though, here's my analysis of the conference parameters you were > asking about: > > mute-detect-sound > Different sound for if muting using only when mute-detect flag is on. > max-members > Specifies the maximum number of participants in a call. > max-members-sound > If caller cannot join because max is reached, this sound plays. Recommended > if max-members is set. > comfort-noise-level > Sets volume of background white noise to generate. > announce-count > Requires TTS. When joining, tells caller how many callers are already in > conference if at least the specified minimum. > suppress-events > ? Sets a flag, but does not appear to do anything with it. > I think suppress-events is to supress audio events such as "User XXXX has joined the conferece." Would the sound right? > > verbose-events > Maximum verbosity for transcripting. > timer-name > Specifies the name of this profile's timer. To separate it from other > timers? > > BB > On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier wrote: > >> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I >> chose flite), uncommenting a few things, and setting the TTS variables in >> the conference profile. You do have to rebuild FS to do this. >> >> With that working, my count function works, too. >> >> I posted a bug last night about conferencing, BTW: if you're using the >> wait-mod function, where the conference doesn't start until the moderator >> arrives, and you're also using separate profiles for users and moderators, >> the users only have the "user" profile until the first moderator arrives. At >> that time, they switch to also be using the "moderator" profile. >> >> Now that TTS is working, I'm going to see about helping you fill out those >> ???s, and maybe see if I can figure out how to fix the above bug. >> >> BB >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/cdc7d183/attachment-0001.html From msc at freeswitch.org Fri Jun 19 13:14:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:14:59 -0500 Subject: [Freeswitch-users] Quick heads up re: minor change in default configs Message-ID: <87f2f3b90906191314j2e301680sd4e0c16e1037cf34@mail.gmail.com> FYI, Just in case this affects you I wanted to give you all a heads up. With the new NAT busting code we found some stuff in the default configs that was no longer applicable and made some changes. Check them out: http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=13874 If you have any dialplan elements that make use of the "use_profile" channel variable please note that we moved it to vars.xml and set it to a default value of 'internal' since that works for just about everything. You can, of course, do whatever you want with it in your customizations. Happy FreeSWITCHing! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/e8fa125e/attachment.html From msc at freeswitch.org Fri Jun 19 13:17:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:17:18 -0500 Subject: [Freeswitch-users] CHANNEL_HANGUP event no longer having variable_billsec in header In-Reply-To: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> References: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> Message-ID: <87f2f3b90906191317n6d47e07du2922a08eb3855b16@mail.gmail.com> Check out this change: http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=13505 Possibly you need to wait for the CHANNEL_HANGUP_COMPLETE event? -MC On Fri, Jun 19, 2009 at 2:15 PM, Matthew Fong wrote: > I upgraded to 13857 today, but noticed that the channel_hangup event no > longer contain the variable_billsec header. > Is this correct, or am I crazy? Thanks. > > --matt > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/eaee4a7d/attachment.html From juanma.v82 at gmail.com Fri Jun 19 13:18:28 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Fri, 19 Jun 2009 17:18:28 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I have another switch who is in charge of it, witch is from another technology), I only want to negotiate the codecs in the way that I want it. This only seams to work when bypass media or proxy media is set to false. Due I need to use it as a SBC(session border controller) or pseudo proxy (I already know that is not intend for it), I need to negotiate the codecs in the FS. In the current thread I've already explained what I'm trying to do. If you give me a tip I'm willing to make the documentation richer. Thanks Regards In my architecture the switch who is in charge of transcoding IS NOT a FS. On 19/06/2009, at 16:19, Brian West wrote: > No right now you can not legally transcode G729 in FreeSWITCH, PERIOD! > > /b > > On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > >> Yes, it can do transcoding. Transcoding isn't the problem to my >> architecture, my problem is the codec negotiation between FS and >> Endpoints. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Fri Jun 19 13:19:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 19 Jun 2009 21:19:33 +0100 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: You are indeed correct, it's the 64bit server that performs well, not the 32bit PAE version. I'm hoping that's the cause. I need to dig around and find out if it's possible to change the kernel remotely and see it sorts the issue. Ultimately I'll update it to 64 bit anyway, but that's a 500 mile trek. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Burgess Sent: 19 June 2009 16:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] high cpu utilization On Fri, Jun 19, 2009 at 2:23 AM, Nik Middleton wrote: > I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I > don't think that's the issue I could be wrong, but I think PAE is a 32-bit kernel adapted for hardware with >4GB RAM. This can create a lot of overhead compared to running a true 64-bit kernel or a 32-bit kernel without PAE. Confirm this with 'uname -a' db _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From matt at hellohunter.com Fri Jun 19 13:21:06 2009 From: matt at hellohunter.com (Matt Hunter) Date: Fri, 19 Jun 2009 13:21:06 -0700 Subject: [Freeswitch-users] CHANNEL_HANGUP event no longer having variable_billsec in header In-Reply-To: <87f2f3b90906191317n6d47e07du2922a08eb3855b16@mail.gmail.com> References: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> <87f2f3b90906191317n6d47e07du2922a08eb3855b16@mail.gmail.com> Message-ID: <4256bf830906191321uf098ff5t62ee353b7c900611@mail.gmail.com> Thanks! On Fri, Jun 19, 2009 at 1:17 PM, Michael Collins wrote: > Check out this change: > http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=13505 > > Possibly you need to wait for the CHANNEL_HANGUP_COMPLETE event? > -MC > > On Fri, Jun 19, 2009 at 2:15 PM, Matthew Fong wrote: > >> I upgraded to 13857 today, but noticed that the channel_hangup event no >> longer contain the variable_billsec header. >> Is this correct, or am I crazy? Thanks. >> >> --matt >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/b6fac9e5/attachment.html From bjbrashier at gmail.com Fri Jun 19 13:20:56 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 13:20:56 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <5a8712120906191305x5a98228eyb1847cfd2905aab5@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> <5a8712120906191305x5a98228eyb1847cfd2905aab5@mail.gmail.com> Message-ID: <7bcfdd290906191320m581dfe57sdb06404bd7e27ce3@mail.gmail.com> That sounds like something it might do, but searching through the code I didn't find any use of the flag that parameter sets. It is, of course, possible that I missed it, especially if it's an indirect use. 2009/6/19 Jo?o Mesquita > Inline... > > On Fri, Jun 19, 2009 at 3:27 PM, Bradley Brashier wrote: > >> So it turns out that it wasn't a bug at all -- it is a feature that was >> not implemented. So I've got some work to do to get that running. Since I >> said I would, though, here's my analysis of the conference parameters you >> were asking about: >> >> mute-detect-sound >> Different sound for if muting using only when mute-detect flag is on. >> max-members >> Specifies the maximum number of participants in a call. >> max-members-sound >> If caller cannot join because max is reached, this sound plays. >> Recommended if max-members is set. >> comfort-noise-level >> Sets volume of background white noise to generate. >> announce-count >> Requires TTS. When joining, tells caller how many callers are already in >> conference if at least the specified minimum. >> suppress-events >> ? Sets a flag, but does not appear to do anything with it. >> > > I think suppress-events is to supress audio events such as "User XXXX has > joined the conferece." > Would the sound right? > > >> >> verbose-events >> Maximum verbosity for transcripting. >> timer-name >> Specifies the name of this profile's timer. To separate it from other >> timers? >> >> BB >> On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier > > wrote: >> >>> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I >>> chose flite), uncommenting a few things, and setting the TTS variables in >>> the conference profile. You do have to rebuild FS to do this. >>> >>> With that working, my count function works, too. >>> >>> I posted a bug last night about conferencing, BTW: if you're using the >>> wait-mod function, where the conference doesn't start until the moderator >>> arrives, and you're also using separate profiles for users and moderators, >>> the users only have the "user" profile until the first moderator arrives. At >>> that time, they switch to also be using the "moderator" profile. >>> >>> Now that TTS is working, I'm going to see about helping you fill out >>> those ???s, and maybe see if I can figure out how to fix the above bug. >>> >>> BB >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/b29dc545/attachment-0001.html From mattdfong at gmail.com Fri Jun 19 13:20:17 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 13:20:17 -0700 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory Message-ID: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> With yesterday's trunk and also a release from 2 weeks ago, I noticed that my freeswitch process as it ran was eating up more and more memory. At the end of the day it was using 75% of the sever's memory (About 12 gigs). It starts out taking a small amount of memory, and then throughout the day it slowly takes more and more. Is this normal? I'm using several lua ivr scripts...and have about 600-900 channels. Whats the best way to go about tracking down the cause? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/d8fabe9a/attachment.html From brian at freeswitch.org Fri Jun 19 13:28:02 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Jun 2009 15:28:02 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> Message-ID: Depends on what you're doing ... or not doing... /b On Jun 19, 2009, at 3:20 PM, Matthew Fong wrote: > With yesterday's trunk and also a release from 2 weeks ago, I > noticed that my freeswitch process as it ran was eating up more and > more memory. At the end of the day it was using 75% of the sever's > memory (About 12 gigs). It starts out taking a small amount of > memory, and then throughout the day it slowly takes more and more. > Is this normal? I'm using several lua ivr scripts...and have about > 600-900 channels. Whats the best way to go about tracking down the > cause? Thanks. > > --matt From anthony.minessale at gmail.com Fri Jun 19 13:38:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Jun 2009 15:38:43 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> Message-ID: <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> are you connecting to a db with the lua? On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: > With yesterday's trunk and also a release from 2 weeks ago, I noticed that > my freeswitch process as it ran was eating up more and more memory. At the > end of the day it was using 75% of the sever's memory (About 12 gigs). It > starts out taking a small amount of memory, and then throughout the day it > slowly takes more and more. Is this normal? I'm using several lua ivr > scripts...and have about 600-900 channels. Whats the best way to go about > tracking down the cause? Thanks. > --matt > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/ab0ccfdc/attachment.html From msc at freeswitch.org Fri Jun 19 13:40:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:40:17 -0500 Subject: [Freeswitch-users] CTI In-Reply-To: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> Message-ID: <87f2f3b90906191340l14e2e0adx2360497b6ee5b464@mail.gmail.com> I'm not aware of any pre-existing software like this. FreeSWITCH has all the hooks for someone to create the abstraction layers for CSTA, TAPI, VoiceXML, etc. but no one has ponied up the money to pay for the development... -MC 2009/6/18 Maxim Tsvetov > Hello! > > We are seeking possibilities to use CTI features with Freeswitch. > > This features are: > - click-to-dial > - call popup > - answer call,hangup > - call transfer > > > Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..) > or there is already written module or third-party software? > This solution should support 100-150 simultaneous ?onnections from > freeswitch users. > > Could you please share you experience with CTI. > > Regards, > Maxim Tsvetov > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/1e2df734/attachment.html From msc at freeswitch.org Fri Jun 19 13:44:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:44:50 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> Message-ID: <87f2f3b90906191344y65a7943bs8c953007f4c95f91@mail.gmail.com> On another email thread Tony mentioned using "top -H" and then doing a gcore to locate the process(es) that are sucking up all the resources. Have you been down that path already? -MC On Fri, Jun 19, 2009 at 3:28 PM, Brian West wrote: > Depends on what you're doing ... or not doing... > > /b > > On Jun 19, 2009, at 3:20 PM, Matthew Fong wrote: > > > With yesterday's trunk and also a release from 2 weeks ago, I > > noticed that my freeswitch process as it ran was eating up more and > > more memory. At the end of the day it was using 75% of the sever's > > memory (About 12 gigs). It starts out taking a small amount of > > memory, and then throughout the day it slowly takes more and more. > > Is this normal? I'm using several lua ivr scripts...and have about > > 600-900 channels. Whats the best way to go about tracking down the > > cause? Thanks. > > > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/6508e66c/attachment.html From mattdfong at gmail.com Fri Jun 19 13:53:57 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 13:53:57 -0700 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> Message-ID: <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> some lua event listeners are connecting to mysql with lua..but the connection is created once, and kept open the lua ivr's do *not *connect to any database. top -H seems to show an even distribution of of cpu and memory usage amongst freeswitch threads. Nothing seems out of the ordinary with a specific thread. --matt On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > are you connecting to a db with the lua? > > > On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: > >> With yesterday's trunk and also a release from 2 weeks ago, I noticed that >> my freeswitch process as it ran was eating up more and more memory. At the >> end of the day it was using 75% of the sever's memory (About 12 gigs). It >> starts out taking a small amount of memory, and then throughout the day it >> slowly takes more and more. Is this normal? I'm using several lua ivr >> scripts...and have about 600-900 channels. Whats the best way to go about >> tracking down the cause? Thanks. >> --matt >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/7f5a0c59/attachment.html From msc at freeswitch.org Fri Jun 19 13:55:04 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:55:04 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> Message-ID: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Now I feel stupid because I didn't read your original post closely enough. You've defined your "mysocket" extension in the "public" context but when you do an origination with sofia/internal/foo at bar it will use the "default" context. I think the quickest way to handle this is to create a copy of your mysocket.xml file and put it in conf/dialplan/default/ and be done with it. -MC On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater wrote: > Mike, > > Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to > me though. > > Max. > > > On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: > >> Max, >> that pastebin failed miserably as none of the xml shows up. can you try >> again or use our pastebin.freeswitch.org site? >> -MC >> >> >> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Hi Mike, >>> >>> It's pasted here: http://pastebin.ca/1466521 >>> >>> Thanks, >>> Max. >>> >>> >>> >>> >>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >>> >>>> Can you turn on debugging (F8) and capture all the output after your >>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>> -MC >>>> >>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>> max.bridgewater at gmail.com> wrote: >>>> >>>>> Any help our there? >>>>> >>>>> I'm still trying to get this piece working. Essentially what i wan to >>>>> do is, when a call comes in (from registered devices as well as unregistered >>>>> devices), notify the my server socket. Somehow it's not working. The change >>>>> i made compared to the standard Freeswitch settings are the following: >>>>> >>>>> 1) Added following extension that in >>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2) Changed file: >>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I noticed that with this extension, all calls received from external >>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>> But calls from registered devices and initiated using the socket interface >>>>> are not forwarded. Is there something that need to be changed in the >>>>> profiles? >>>>> >>>>> or is something wrong with my dial string? >>>>> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >>>>> >>>>> >>>>> In the logs, i cannot see that that my extension is being matched. >>>>> >>>>> Any idea, >>>>> >>>>> Max. >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/84e3ac62/attachment-0001.html From msc at freeswitch.org Fri Jun 19 13:59:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:59:02 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> Message-ID: <87f2f3b90906191359t179fc74bw74e24abca3c4f42c@mail.gmail.com> Can you pastebin your script and your dialplan? There might be some clue there.. On Fri, Jun 19, 2009 at 3:53 PM, Matthew Fong wrote: > some lua event listeners are connecting to mysql with lua..but the > connection is created once, and kept open > the lua ivr's do *not *connect to any database. > > top -H seems to show an even distribution of of cpu and memory usage > amongst freeswitch threads. Nothing seems out of the ordinary with a > specific thread. > > --matt > On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> are you connecting to a db with the lua? >> >> >> On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: >> >>> With yesterday's trunk and also a release from 2 weeks ago, I noticed >>> that my freeswitch process as it ran was eating up more and more memory. At >>> the end of the day it was using 75% of the sever's memory (About 12 gigs). >>> It starts out taking a small amount of memory, and then throughout the day >>> it slowly takes more and more. Is this normal? I'm using several lua ivr >>> scripts...and have about 600-900 channels. Whats the best way to go about >>> tracking down the cause? Thanks. >>> --matt >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/762d3a78/attachment.html From anthony.minessale at gmail.com Fri Jun 19 14:07:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Jun 2009 16:07:37 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> Message-ID: <191c3a030906191407v1eab84bes7f61704a2ef77e4@mail.gmail.com> try removing certian elements of you setup to narrrow it down one at a time. remove the lua + sql, the ivr scripts etc and see if you can pinpoint your problem it's amost for sure going to be in lua and probably some plugin for it. On Fri, Jun 19, 2009 at 3:53 PM, Matthew Fong wrote: > some lua event listeners are connecting to mysql with lua..but the > connection is created once, and kept open > the lua ivr's do *not *connect to any database. > > top -H seems to show an even distribution of of cpu and memory usage > amongst freeswitch threads. Nothing seems out of the ordinary with a > specific thread. > > --matt > On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> are you connecting to a db with the lua? >> >> >> On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: >> >>> With yesterday's trunk and also a release from 2 weeks ago, I noticed >>> that my freeswitch process as it ran was eating up more and more memory. At >>> the end of the day it was using 75% of the sever's memory (About 12 gigs). >>> It starts out taking a small amount of memory, and then throughout the day >>> it slowly takes more and more. Is this normal? I'm using several lua ivr >>> scripts...and have about 600-900 channels. Whats the best way to go about >>> tracking down the cause? Thanks. >>> --matt >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/29fb08d4/attachment.html From apt.get at gmail.com Fri Jun 19 14:09:13 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 19 Jun 2009 15:09:13 -0600 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: On Fri, Jun 19, 2009 at 2:19 PM, Nik Middleton wrote: > You are indeed correct, it's the 64bit server that performs well, not > the 32bit PAE version. ?I'm hoping that's the cause. ?I need to dig > around and find out if it's possible to change the kernel remotely and > see it sorts the issue. ?Ultimately I'll update it to 64 bit anyway, but > that's a 500 mile trek. I don't think it's as simple as changing the kernel. My understanding is that when you change arch you are reinstalling the system. Given that 32-bit binaries and libs generally will run in a 64-bit environment though, there may be a way to swing it. I'm not sure how you would determine whether that were really your problem. No question a PAE kernel does create overhead though, in some situations more than others. db From max.bridgewater at gmail.com Fri Jun 19 14:22:33 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 17:22:33 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Message-ID: I don't have my settings to try it right now. Still i have a question. If it's the way you describe it, why wouldn't sofia/extenal/foo at bar solve the problem? I think i even copied the extension both to the default directory. But i will confirm and let you know. Max. On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins wrote: > Now I feel stupid because I didn't read your original post closely enough. > > > You've defined your "mysocket" extension in the "public" context but when > you do an origination with sofia/internal/foo at bar it will use the > "default" context. I think the quickest way to handle this is to create a > copy of your mysocket.xml file and put it in conf/dialplan/default/ and be > done with it. > > -MC > > > On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Mike, >> >> Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to >> me though. >> >> Max. >> >> >> On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: >> >>> Max, >>> that pastebin failed miserably as none of the xml shows up. can you try >>> again or use our pastebin.freeswitch.org site? >>> -MC >>> >>> >>> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >>> max.bridgewater at gmail.com> wrote: >>> >>>> Hi Mike, >>>> >>>> It's pasted here: http://pastebin.ca/1466521 >>>> >>>> Thanks, >>>> Max. >>>> >>>> >>>> >>>> >>>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >>>> >>>>> Can you turn on debugging (F8) and capture all the output after your >>>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>>> -MC >>>>> >>>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>>> max.bridgewater at gmail.com> wrote: >>>>> >>>>>> Any help our there? >>>>>> >>>>>> I'm still trying to get this piece working. Essentially what i wan to >>>>>> do is, when a call comes in (from registered devices as well as unregistered >>>>>> devices), notify the my server socket. Somehow it's not working. The change >>>>>> i made compared to the standard Freeswitch settings are the following: >>>>>> >>>>>> 1) Added following extension that in >>>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2) Changed file: >>>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> I noticed that with this extension, all calls received from external >>>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>>> But calls from registered devices and initiated using the socket interface >>>>>> are not forwarded. Is there something that need to be changed in the >>>>>> profiles? >>>>>> >>>>>> or is something wrong with my dial string? >>>>>> {origination_caller_id_number=12000}sofia/internal/ >>>>>> 242424 at 192.168.1.62. >>>>>> >>>>>> In the logs, i cannot see that that my extension is being matched. >>>>>> >>>>>> Any idea, >>>>>> >>>>>> Max. >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/5b80a16d/attachment-0001.html From timb0311 at hotmail.com Fri Jun 19 17:52:31 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 20:52:31 -0400 Subject: [Freeswitch-users] Update - Transmit fax locally for test In-Reply-To: References: Message-ID: Yes that is the extension defined in the default dialplan. It is setup like explained under mod_fax... here is the actual definition: http://pastebin.freeswitch.org/9450 > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503 at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000 at x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users at lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > _________________________________________________________________ Hotmail? has ever-growing storage! Don?t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/9ab0f893/attachment.html From timb0311 at hotmail.com Fri Jun 19 20:06:10 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 23:06:10 -0400 Subject: [Freeswitch-users] (Found Fix) Transmit fax locally for test In-Reply-To: References: Message-ID: Ok so after many attempts of trial and error I narrowed it down to acls. So when trying to orginate a call to the local FS extension it was getting blocked. Adding the following allow with my freeswitch IP to the domains list allowed the originate to take place. acl.conf.xml: So now this statement works for local fax testing: originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) Now my question is, is this the proper or best way to configure this? Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503 at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000 at x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users at lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > > haha, yeah i see it now... duh. pulled an all nighter, too many things > going on. must have overlooked it. > > > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > > the login and password is pastebin/freeswitch > > > > been trying to break myself into freeswitch on top of my original workload. > thanks for the help. > _________________________________________________________________ Bing? brings you maps, menus, and reviews organized in one place. Try it now. http://www.bing.com/search?q=restaurants&form=MLOGEN&publ=WLHMTAG&crea=TEXT_MLOGEN_Core_tagline_local_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/f05205dd/attachment.html From mariano.dellano at gmail.com Fri Jun 19 14:28:04 2009 From: mariano.dellano at gmail.com (Mariano de Llano) Date: Fri, 19 Jun 2009 18:28:04 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: You are right, it seams that it can not be done. In the past I've tried to do something similiar but with no success. Apperently the documentation is wrong. If I have some time I will look at the code and I will give you some feedback. Cheers, PS: The transcoding question has nothing to do with your question :) On 19/06/2009, at 17:18, JuanMa wrote: > I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I > have another switch who is in charge of it, witch is from another > technology), I only want to negotiate the codecs in the way that I > want it. This only seams to work when bypass media or proxy media is > set to false. Due I need to use it as a SBC(session border controller) > or pseudo proxy (I already know that is not intend for it), I need to > negotiate the codecs in the FS. In the current thread I've already > explained what I'm trying to do. If you give me a tip I'm willing to > make the documentation richer. > > Thanks > Regards > > In my architecture the switch who is in charge of transcoding IS NOT a > FS. > > > On 19/06/2009, at 16:19, Brian West wrote: > >> No right now you can not legally transcode G729 in FreeSWITCH, >> PERIOD! >> >> /b >> >> On Jun 19, 2009, at 2:11 PM, JuanMa wrote: >> >>> Yes, it can do transcoding. Transcoding isn't the problem to my >>> architecture, my problem is the codec negotiation between FS and >>> Endpoints. >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Jun 19 21:44:30 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 20 Jun 2009 00:44:30 -0400 Subject: [Freeswitch-users] Mod_Fax / TxFax / Originate In-Reply-To: References: Message-ID: <6D803586-B8E1-4EAB-A780-6A3EAD680C5F@jerris.com> Try using loopback endpoint for this test . Mike On Jun 17, 2009, at 10:00 AM, Tim B wrote: > Trying to do a local test for faxing. Keep getting an error. > > default dialplan: > > > > > > > > > > > //inbound from remote box works fine > - connect asterisk box and fs box, then fax from asterisk to fs... OK > - also fax from fs to asterisk.... OK > > // local fax on fs .... FAILS!! > // my originate command: > originate sofia/internal/8000 at 192.168.10.35 &txfax(storage/fax/ > test.tif) > > // error as follows: > 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing > FreeSWITCH->8000 in context public > 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer > 1 Legged calls > 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at 192.168.10.35 > [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > > > > > Lauren found her dream laptop. Find the PC that?s right for you. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/d21742bc/attachment-0001.html From mike at jerris.com Fri Jun 19 21:46:59 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 20 Jun 2009 00:46:59 -0400 Subject: [Freeswitch-users] Freeswitch as a B2B Application Server for IMS In-Reply-To: References: Message-ID: Some of these things make sense is some scenarios but not others. Most people are wanting to do full topology hiding, so we don't by default pass very much across a bridge. I am interested in working on this, feel free to contact me off list with your findings. Mike On Jun 17, 2009, at 11:20 AM, Cavalera Claudio Luigi wrote: > Hello freeswitchers, > lately I'm trying to set up a testbed to ivestigate a potential use of > freeswitch as a Back-to-Back application server in an IMS > architecture. > > I've seen IMS specs are also linked here > http://wiki.freeswitch.org/wiki/Documentation > so I've thinked maybe there's a chance :-) > > I also have the inner feeling that fs could do an amazing job in IMS > field as he does in NGN. > > For my testing I'm now using OpenIMSCore as control layer (where > phones > do register) and I'm trying to put fs on top of it as a b2b > application > server to provide services. > > I would like to share with you my experience to see if something could > be done about this scenario (or if it's worth the trouble at least). > > Alice and Bob are two users registered to the IMS Core and they both > have a profile for which originating and terminating INVITEs get > triggered towards the fs application server. > > When Alice calls Bob the call setup would include three legs: > > 1) Alice -> PCSCF (orig) -> SCSCF (orig) -> FS (orig) > > 2) FS (orig) -> SCSCF (orig) -> ICSCF -> SCSCF (term) -> FS (term) > > 3) FS (term) -> SCSCF (term) -> PCSCF(term) -> Bob > > > This partially works already with fs out of the box, but there are a > still a few issues to be solved. > > When FS starts the brand new leg 2) as a B2B User Agent he should keep > the Route: header in the SIP INVITE "almost the same" as the one he > received in the leg 1) > > I see here two different issues: > a) Getting the Route: header out of incoming invite in leg 1) > b) Writing the proper Route: header and have FS behaving correctly at > transport layer in the outgoing INVITE in leg 2) > > a) Now please correct me if I'm wrong: at the moment the > header > is not a channel variable available in fs (e.g. I don't get it with > the > "info" app). It there were a way to get this header out of the > incoming > INVITEs, I could do the logic to parse it and forge a proper one in > the > outgoing INVITE. > > > b) Concerning how to write the header, I'm already working > with > fs_path directive which also makes FS behaves correctly at network > layer. > Could someone please elaborate a little bit about the alternative to > fs_path directive? > > I've seen there are already many in theory: > - combining sip_h_Route= with > http://wiki.freeswitch.org/wiki/Variable_sip_network_destination > - use of http://wiki.freeswitch.org/wiki/Variable_sip_route_uri > - use of fs_path= > http://wiki.freeswitch.org/wiki/ > Sofia#Specifying_SIP_Proxy_With_fs_path > > > I've simplified the scenario a little bit, there are other things that > the B2B AS should do (e.g. removing Record-Route:) but FS do them > already from what I've tested. > If anyone in the community is interested I'm here to provide further > information or share my experience to implement the best solution. > > Best regards, > Claudio > > > > > Internet Email Confidentiality Footer > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e > ogni documento o file allegato, e' rivolta unicamente alla/e persona/ > e cui e' indirizzata ed alle altre da questa autorizzata/e a > riceverla. Se non siete i destinatari/autorizzati siete avvisati che > qualsiasi azione, copia, comunicazione, divulgazione o simili basate > sul contenuto di tali informazioni e' vietata e potrebbe essere > contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia > di protezione dei dati personali). Se avete ricevuto questa > comunicazione per errore, vi preghiamo di darne immediata notizia al > mittente e di distruggere il messaggio originale e ogni file > allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) > only and are confidential and/or may contain legally privileged > information. If you have received this message by mistake or are not > one of the addressees above, you may take no action based on it, and > you may not copy or show it to anyone; please reply to this e-mail > and point out the error which has occurred. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bipin at xbipin.com Fri Jun 19 22:42:08 2009 From: bipin at xbipin.com (xbipin) Date: Fri, 19 Jun 2009 22:42:08 -0700 (PDT) Subject: [Freeswitch-users] need to get some basics right Message-ID: <24122294.post@talk.nabble.com> im just confused about certains tuff in freeswitch, firstly if we use TLS for SIP then FS can read such packets but if it were to act as gateway can it communicate in plain SIP, so its like phone connected to FS using TLS and then from FS to gateway in plain SIP? secondly, if phone uses SRTP, can FS take that media stream and convert to plain RTP so it can forward it to remote Gateway, i know it cant do for G723, G729 due to license issues but can it do for the other codecs? my main concern is can it actually act as a encryption and decryption server by remaining between the gateway and phone? -- View this message in context: http://www.nabble.com/need-to-get-some-basics-right-tp24122294p24122294.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mrene_lists at avgs.ca Fri Jun 19 22:44:28 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 20 Jun 2009 00:44:28 -0500 Subject: [Freeswitch-users] need to get some basics right In-Reply-To: <24122294.post@talk.nabble.com> References: <24122294.post@talk.nabble.com> Message-ID: You can do all that. SRTP<>RTP doesnt depend on the codec since it just decrypts the packet's contents and SIP/TLS is dependent on that specific connection. Math On 20-Jun-09, at 12:42 AM, xbipin wrote: > > im just confused about certains tuff in freeswitch, firstly if we > use TLS for > SIP then FS can read such packets but if it were to act as gateway > can it > communicate in plain SIP, so its like phone connected to FS using > TLS and > then from FS to gateway in plain SIP? > secondly, if phone uses SRTP, can FS take that media stream and > convert to > plain RTP so it can forward it to remote Gateway, i know it cant do > for > G723, G729 due to license issues but can it do for the other codecs? > > my main concern is can it actually act as a encryption and > decryption server > by remaining between the gateway and phone? > -- > View this message in context: http://www.nabble.com/need-to-get-some-basics-right-tp24122294p24122294.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raffaele.p.guidi at gmail.com Sat Jun 20 05:40:56 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sat, 20 Jun 2009 14:40:56 +0200 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3AF28D.4040708@gmail.com> References: <4A3AF28D.4040708@gmail.com> Message-ID: uhm... nice! But why not MPL license (the same as freeswitch)? On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com wrote: > http://versafon.com/versafonweb/Software.jsp > > Essentially it's a wrapper around inbound socket interface, not all > events supported yet, and not all event parameters/variables. It's multi > threaded and scaled well in testing. > We offer commercial support and development for FreeSwitch as well. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/defe058c/attachment.html From diego.viola at gmail.com Sat Jun 20 06:00:37 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 20 Jun 2009 09:00:37 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: References: <4A3AF28D.4040708@gmail.com> Message-ID: <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> Because they probably want a stronger copyleft license? I prefer the GPL because of that reason. On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > uhm... nice! But why not MPL license (the same as freeswitch)? > > > On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com wrote: > >> http://versafon.com/versafonweb/Software.jsp >> >> Essentially it's a wrapper around inbound socket interface, not all >> events supported yet, and not all event parameters/variables. It's multi >> threaded and scaled well in testing. >> We offer commercial support and development for FreeSwitch as well. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/7c39acb0/attachment.html From paul.degt at gmail.com Sat Jun 20 07:37:59 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 10:37:59 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> Message-ID: <4A3CF447.3030501@gmail.com> Yes, that's one of the reasons. Another point is that GPL v.3 is defined more clearly from legal perspective, at least from our legal adviser point of view. Diego Viola wrote: > Because they probably want a stronger copyleft license? > > I prefer the GPL because of that reason. > > On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > > wrote: > > uhm... nice! But why not MPL license (the same as freeswitch)? > > > On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > > wrote: > > http://versafon.com/versafonweb/Software.jsp > > Essentially it's a wrapper around inbound socket interface, > not all > events supported yet, and not all event parameters/variables. > It's multi > threaded and scaled well in testing. > We offer commercial support and development for FreeSwitch as > well. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From saeedahmad1981 at gmail.com Sat Jun 20 08:06:35 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Sat, 20 Jun 2009 17:06:35 +0200 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: Hi, I am not expert, and if i understood you correctly, i am doing it like that: set late negotiation to false (it can be set true if you want to negotiate it on the fly, but it didn't work for me) don't use absolute_codec_string instead use codec_string and put codec(s) what EP2 is offering, now if A side is sending one of the codec which are defined for EP2 then it will be used. i didn't consider your FS at right side because your SWITCH definately going to transcode the codec into G729, otherwise above scenario should be same for both FS. let me know if it also works for you, i am also using it in proxy media mode. are you using xml_curl? - Saeed On Fri, Jun 19, 2009 at 11:28 PM, Mariano de Llano < mariano.dellano at gmail.com> wrote: > > You are right, it seams that it can not be done. > > In the past I've tried to do something similiar but with no success. > Apperently the documentation is wrong. > > If I have some time I will look at the code and I will give you some > feedback. > > Cheers, > > PS: The transcoding question has nothing to do with your question :) > > On 19/06/2009, at 17:18, JuanMa wrote: > > > I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I > > have another switch who is in charge of it, witch is from another > > technology), I only want to negotiate the codecs in the way that I > > want it. This only seams to work when bypass media or proxy media is > > set to false. Due I need to use it as a SBC(session border controller) > > or pseudo proxy (I already know that is not intend for it), I need to > > negotiate the codecs in the FS. In the current thread I've already > > explained what I'm trying to do. If you give me a tip I'm willing to > > make the documentation richer. > > > > Thanks > > Regards > > > > In my architecture the switch who is in charge of transcoding IS NOT a > > FS. > > > > > > On 19/06/2009, at 16:19, Brian West wrote: > > > >> No right now you can not legally transcode G729 in FreeSWITCH, > >> PERIOD! > >> > >> /b > >> > >> On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > >> > >>> Yes, it can do transcoding. Transcoding isn't the problem to my > >>> architecture, my problem is the codec negotiation between FS and > >>> Endpoints. > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/98688012/attachment-0001.html From brian at freeswitch.org Sat Jun 20 08:26:27 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 20 Jun 2009 10:26:27 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3CF447.3030501@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> Message-ID: <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> I still say why not MPL or at the very least MPL/GPL? /b On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > Yes, that's one of the reasons. Another point is that GPL v.3 is > defined > more clearly from legal perspective, at least from our legal adviser > point of view. > > Diego Viola wrote: >> Because they probably want a stronger copyleft license? >> >> I prefer the GPL because of that reason. >> >> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi >> > >> wrote: >> >> uhm... nice! But why not MPL license (the same as freeswitch)? >> >> >> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com >> > > wrote: >> >> http://versafon.com/versafonweb/Software.jsp >> >> Essentially it's a wrapper around inbound socket interface, >> not all >> events supported yet, and not all event parameters/variables. >> It's multi >> threaded and scaled well in testing. >> We offer commercial support and development for FreeSwitch as >> well. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul.degt at gmail.com Sat Jun 20 09:12:15 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 12:12:15 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> Message-ID: <4A3D0A5F.4080608@gmail.com> As I mentioned the decision was influenced by our legal adviser. And it probably can be relicensed as MPL/GPL, but why? My understanding is that for an end user which wants to use a free open source software there should be absolutely no difference between GPL and MPL. But if any one has has any licensing concern with his particular use case - please contact me with the details, I will do what I can to help. Brian West wrote: > I still say why not MPL or at the very least MPL/GPL? > > /b > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is >> defined >> more clearly from legal perspective, at least from our legal adviser >> point of view. >> >> Diego Viola wrote: >> >>> Because they probably want a stronger copyleft license? >>> >>> I prefer the GPL because of that reason. >>> >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi >>> > >>> wrote: >>> >>> uhm... nice! But why not MPL license (the same as freeswitch)? >>> >>> >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com >>> >> > wrote: >>> >>> http://versafon.com/versafonweb/Software.jsp >>> >>> Essentially it's a wrapper around inbound socket interface, >>> not all >>> events supported yet, and not all event parameters/variables. >>> It's multi >>> threaded and scaled well in testing. >>> We offer commercial support and development for FreeSwitch as >>> well. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sat Jun 20 09:29:21 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 20 Jun 2009 11:29:21 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D0A5F.4080608@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> Message-ID: <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> There actually are issues between the GPL and MPL :P /b On Jun 20, 2009, at 11:12 AM, paul.degt wrote: > source software there should be absolutely no difference between GPL > and > MPL. From paul.degt at gmail.com Sat Jun 20 10:07:18 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 13:07:18 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> Message-ID: <4A3D1746.9050504@gmail.com> Not unless you combining GPL and MPL code in one binary, which I find highly improbable in case of FreeSwitch. And even in this case there seems to be a workaround: "However, MPL 1.1 has a provision (section 13) that allows a program (or parts of it) to offer a choice of another license as well. If part of a program allows the GNU GPL as an alternate choice, or any other GPL-compatible license as an alternate choice, that part of the program has a GPL-compatible license." http://www.fsf.org/licensing/licenses/index_html#GPLIncompatibleLicenses Brian West wrote: > There actually are issues between the GPL and MPL :P > > /b > > On Jun 20, 2009, at 11:12 AM, paul.degt wrote: > > >> source software there should be absolutely no difference between GPL >> and >> MPL. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Sat Jun 20 10:21:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 20 Jun 2009 12:21:15 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D1746.9050504@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> <4A3D1746.9050504@gmail.com> Message-ID: <191c3a030906201021o74aa892at68bef899089bd847@mail.gmail.com> License is less important in a socket interface lib. Its not really a big deal what license it is guys. However, I did create a BSD ESL lib distributed with FS to make sure nobody would have any license problems connecting to our stuff. If you don't like this lib consider using swig on the wrapper lib in libesl like I have done for perl, lua, php, python, ruby etc. On Jun 20, 2009 12:15 PM, "paul.degt" wrote: Not unless you combining GPL and MPL code in one binary, which I find highly improbable in case of FreeSwitch. And even in this case there seems to be a workaround: "However, MPL 1.1 has a provision (section 13) that allows a program (or parts of it) to offer a choice of another license as well. If part of a program allows the GNU GPL as an alternate choice, or any other GPL-compatible license as an alternate choice, that part of the program has a GPL-compatible license." http://www.fsf.org/licensing/licenses/index_html#GPLIncompatibleLicenses Brian West wrote: > There actually are issues between the GPL and MPL :P > > /b > > On Jun 20, 200... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/99398f40/attachment.html From d at d-man.org Sat Jun 20 12:32:31 2009 From: d at d-man.org (Darren Schreiber) Date: Sat, 20 Jun 2009 12:32:31 -0700 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 Message-ID: <2C6A726363C34E2DAE22EF723C668E1C@test> Hello folks, An important stability issue was identified in the FreeSWITCH core ODBC drivers when utilizing decimal / float columns in databases (at least MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you are likely using floating point or decimal columns to track cash amounts. FreeSWITCH may periodically core dump when a floating point value is retrieved from the database due to this bug. Please upgrade to rev 13866 (or at least apply the patch - it can be applied independently if you are on at least rev 12152). You do not need to update mod_nibblebill, as the bug is not actually in mod_nibblebill, it just happens to cause the right conditions to exist to exhibit this behavior. Background information is in FSCORE-384 . Thanks to Tony for acting quickly on this patch. - Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/9dd200b0/attachment.html From raffaele.p.guidi at gmail.com Sat Jun 20 12:39:34 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sat, 20 Jun 2009 21:39:34 +0200 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D0A5F.4080608@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> Message-ID: Well, GPL for a library is a dead end. Nobody will use it unless for open source projects. I would consider any possible choice before using it. Regards, Raffaele On Sat, Jun 20, 2009 at 18:12, paul.degt wrote: > As I mentioned the decision was influenced by our legal adviser. And it > probably can be relicensed as MPL/GPL, but why? > My understanding is that for an end user which wants to use a free open > source software there should be absolutely no difference between GPL and > MPL. > But if any one has has any licensing concern with his particular use > case - please contact me with the details, I will do what I can to help. > > > Brian West wrote: > > I still say why not MPL or at the very least MPL/GPL? > > > > /b > > > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > > > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is > >> defined > >> more clearly from legal perspective, at least from our legal adviser > >> point of view. > >> > >> Diego Viola wrote: > >> > >>> Because they probably want a stronger copyleft license? > >>> > >>> I prefer the GPL because of that reason. > >>> > >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > >>> > > >>> wrote: > >>> > >>> uhm... nice! But why not MPL license (the same as freeswitch)? > >>> > >>> > >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > >>> >>> > wrote: > >>> > >>> http://versafon.com/versafonweb/Software.jsp > >>> > >>> Essentially it's a wrapper around inbound socket interface, > >>> not all > >>> events supported yet, and not all event parameters/variables. > >>> It's multi > >>> threaded and scaled well in testing. > >>> We offer commercial support and development for FreeSwitch as > >>> well. > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > ------------------------------------------------------------------------ > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/321913f5/attachment-0001.html From diego.viola at gmail.com Sat Jun 20 12:41:11 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 20 Jun 2009 15:41:11 -0400 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 In-Reply-To: <2C6A726363C34E2DAE22EF723C668E1C@test> References: <2C6A726363C34E2DAE22EF723C668E1C@test> Message-ID: <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> Is this issue fixed in latest trunk? On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber wrote: > Hello folks, > An important stability issue was identified in the FreeSWITCH core > ODBC drivers when utilizing decimal / float columns in databases (at least > MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you > are likely using floating point or decimal columns to track cash amounts. > FreeSWITCH may periodically core dump when a floating point value is > retrieved from the database due to this bug. > > Please upgrade to rev 13866 (or at least apply the patch - > it can be applied independently if you are on at least rev 12152). You do > not need to update mod_nibblebill, as the bug is not actually in > mod_nibblebill, it just happens to cause the right conditions to exist to > exhibit this behavior. > > Background information is in FSCORE-384 > . > > Thanks to Tony for acting quickly on this patch. > > - Darren > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/3fdb9ffe/attachment.html From d at d-man.org Sat Jun 20 12:44:26 2009 From: d at d-man.org (Darren Schreiber) Date: Sat, 20 Jun 2009 12:44:26 -0700 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 In-Reply-To: <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> References: <2C6A726363C34E2DAE22EF723C668E1C@test> <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> Message-ID: <113AC34E3E23432D8B7F87FAC3CC40A2@test> Yes _____ From: Diego Viola [mailto:diego.viola at gmail.com] Sent: Saturday, June 20, 2009 12:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] If you use mod_nibblebill,please upgrade to rev 13866 Is this issue fixed in latest trunk? On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber wrote: Hello folks, An important stability issue was identified in the FreeSWITCH core ODBC drivers when utilizing decimal / float columns in databases (at least MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you are likely using floating point or decimal columns to track cash amounts. FreeSWITCH may periodically core dump when a floating point value is retrieved from the database due to this bug. Please upgrade to rev 13866 (or at least apply the patch - it can be applied independently if you are on at least rev 12152). You do not need to update mod_nibblebill, as the bug is not actually in mod_nibblebill, it just happens to cause the right conditions to exist to exhibit this behavior. Background information is in FSCORE-384 . Thanks to Tony for acting quickly on this patch. - Darren _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/8283ad13/attachment.html From diego.viola at gmail.com Sat Jun 20 12:45:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 20 Jun 2009 15:45:47 -0400 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 In-Reply-To: <113AC34E3E23432D8B7F87FAC3CC40A2@test> References: <2C6A726363C34E2DAE22EF723C668E1C@test> <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> <113AC34E3E23432D8B7F87FAC3CC40A2@test> Message-ID: <86a32abc0906201245r43de8aboe17103541fab9682@mail.gmail.com> Cool, thanks, you guys are fast like the light ;) On Sat, Jun 20, 2009 at 3:44 PM, Darren Schreiber wrote: > Yes > > ------------------------------ > *From:* Diego Viola [mailto:diego.viola at gmail.com] > *Sent:* Saturday, June 20, 2009 12:41 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] If you use mod_nibblebill,please upgrade > to rev 13866 > > Is this issue fixed in latest trunk? > > On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber wrote: > >> Hello folks, >> An important stability issue was identified in the FreeSWITCH core >> ODBC drivers when utilizing decimal / float columns in databases (at least >> MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you >> are likely using floating point or decimal columns to track cash amounts. >> FreeSWITCH may periodically core dump when a floating point value is >> retrieved from the database due to this bug. >> >> Please upgrade to rev 13866 (or at least apply the patch - >> it can be applied independently if you are on at least rev 12152). You do >> not need to update mod_nibblebill, as the bug is not actually in >> mod_nibblebill, it just happens to cause the right conditions to exist to >> exhibit this behavior. >> >> Background information is in FSCORE-384 >> . >> >> Thanks to Tony for acting quickly on this patch. >> >> - Darren >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/58bc1057/attachment.html From evilla at chipoly.com Sat Jun 20 12:51:48 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Sat, 20 Jun 2009 13:51:48 -0600 Subject: [Freeswitch-users] stucked with mod_fsv and mod_h26x Message-ID: <009001c9f1e0$8d1df990$a759ecb0$@com> Hello friends. i'm kind of stucked with mod_fsv and mod_h26x... there is no info on wiki yet. have u used them? Can u send examples of how-to? I need to join 2 endpoints with full HD video conference using PZT HD VideoCams to PC and SIP-UA. Thank you! EDWIN ChiPoLy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/14cf2161/attachment.html From paul.degt at gmail.com Sat Jun 20 13:15:40 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 16:15:40 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> Message-ID: <4A3D436C.6030405@gmail.com> That's rather bold statement and indicates lack of knowledge on the subject . I would advise you to look at http://www.fsf.org/licensing/licenses/gpl-faq.html on GPL information. In fact proprietary applications can use/modify GPL software with no limitations or obligations towards authors. The only requirement is if such applications will be published/distributed it must be done under GPL license. If that's a problem - just ask authors to release a commercial version for such purpose. I think GPL is fair and easy to deal with. Raffaele P. Guidi wrote: > Well, GPL for a library is a dead end. Nobody will use it unless for > open source projects. I would consider any possible choice before > using it. > > Regards, > Raffaele > > On Sat, Jun 20, 2009 at 18:12, paul.degt > wrote: > > As I mentioned the decision was influenced by our legal adviser. > And it > probably can be relicensed as MPL/GPL, but why? > My understanding is that for an end user which wants to use a free > open > source software there should be absolutely no difference between > GPL and > MPL. > But if any one has has any licensing concern with his particular use > case - please contact me with the details, I will do what I can to > help. > > > Brian West wrote: > > I still say why not MPL or at the very least MPL/GPL? > > > > /b > > > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > > > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is > >> defined > >> more clearly from legal perspective, at least from our legal > adviser > >> point of view. > >> > >> Diego Viola wrote: > >> > >>> Because they probably want a stronger copyleft license? > >>> > >>> I prefer the GPL because of that reason. > >>> > >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > >>> > >> > >>> wrote: > >>> > >>> uhm... nice! But why not MPL license (the same as freeswitch)? > >>> > >>> > >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > > >>> > > > >>> >> > wrote: > >>> > >>> http://versafon.com/versafonweb/Software.jsp > >>> > >>> Essentially it's a wrapper around inbound socket interface, > >>> not all > >>> events supported yet, and not all event > parameters/variables. > >>> It's multi > >>> threaded and scaled well in testing. > >>> We offer commercial support and development for > FreeSwitch as > >>> well. > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> > > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > ------------------------------------------------------------------------ > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From raffaele.p.guidi at gmail.com Sat Jun 20 13:49:47 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sat, 20 Jun 2009 22:49:47 +0200 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D436C.6030405@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <4A3D436C.6030405@gmail.com> Message-ID: Right, fair and easy to deal with. And also to NOT deal with - as would any developer working in a commercial company. Again, I would consider any possible choice before using it. Regards, Raffaele On Sat, Jun 20, 2009 at 22:15, paul.degt wrote: > That's rather bold statement and indicates lack of knowledge on the > subject . I would advise you to look at > http://www.fsf.org/licensing/licenses/gpl-faq.html on GPL information. > In fact proprietary applications can use/modify GPL software with no > limitations or obligations towards authors. The only requirement is if > such applications will be published/distributed it must be done under > GPL license. If that's a problem - just ask authors to release a > commercial version for such purpose. > I think GPL is fair and easy to deal with. > > > Raffaele P. Guidi wrote: > > Well, GPL for a library is a dead end. Nobody will use it unless for > > open source projects. I would consider any possible choice before > > using it. > > > > Regards, > > Raffaele > > > > On Sat, Jun 20, 2009 at 18:12, paul.degt > > wrote: > > > > As I mentioned the decision was influenced by our legal adviser. > > And it > > probably can be relicensed as MPL/GPL, but why? > > My understanding is that for an end user which wants to use a free > > open > > source software there should be absolutely no difference between > > GPL and > > MPL. > > But if any one has has any licensing concern with his particular use > > case - please contact me with the details, I will do what I can to > > help. > > > > > > Brian West wrote: > > > I still say why not MPL or at the very least MPL/GPL? > > > > > > /b > > > > > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > > > > > > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is > > >> defined > > >> more clearly from legal perspective, at least from our legal > > adviser > > >> point of view. > > >> > > >> Diego Viola wrote: > > >> > > >>> Because they probably want a stronger copyleft license? > > >>> > > >>> I prefer the GPL because of that reason. > > >>> > > >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > > >>> > > > > >> > > >>> wrote: > > >>> > > >>> uhm... nice! But why not MPL license (the same as freeswitch)? > > >>> > > >>> > > >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > > > > >>> > > > > > >>> >> > > wrote: > > >>> > > >>> http://versafon.com/versafonweb/Software.jsp > > >>> > > >>> Essentially it's a wrapper around inbound socket > interface, > > >>> not all > > >>> events supported yet, and not all event > > parameters/variables. > > >>> It's multi > > >>> threaded and scaled well in testing. > > >>> We offer commercial support and development for > > FreeSwitch as > > >>> well. > > >>> > > >>> _______________________________________________ > > >>> Freeswitch-users mailing list > > >>> Freeswitch-users at lists.freeswitch.org > > > > >>> > > > > >>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >>> > > >>> _______________________________________________ > > >>> Freeswitch-users mailing list > > >>> Freeswitch-users at lists.freeswitch.org > > > > >>> > > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >>> > > > ------------------------------------------------------------------------ > > >>> > > >>> _______________________________________________ > > >>> Freeswitch-users mailing list > > >>> Freeswitch-users at lists.freeswitch.org > > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/3d05fdc8/attachment.html From anthony.minessale at gmail.com Sat Jun 20 15:10:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 20 Jun 2009 17:10:22 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <4A3D436C.6030405@gmail.com> Message-ID: <191c3a030906201510t784275b4j17173edc5684a7ab@mail.gmail.com> We don't need to argue licenses the topic is way too subjective. To each his own. Let's concentrate on the code and the cool features we can make in the future. On Jun 20, 2009 4:00 PM, "Raffaele P. Guidi" wrote: Right, fair and easy to deal with. And also to NOT deal with - as would any developer working in a commercial company. Again, I would consider any possible choice before using it. Regards, Raffaele On Sat, Jun 20, 2009 at 22:15, paul.degt wrote: > > That's rather bold statem... _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/b72004b7/attachment.html From steveu at coppice.org Sat Jun 20 17:57:50 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 21 Jun 2009 08:57:50 +0800 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3CF447.3030501@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> Message-ID: <4A3D858E.5050203@coppice.org> paul.degt wrote: > Yes, that's one of the reasons. Another point is that GPL v.3 is defined > more clearly from legal perspective, at least from our legal adviser > point of view. > While the legal status of MPL is widely considered to be vague, is GPL 3 any better? GPL 2 is pretty sound, and has stood the test of time. However a number of large companies have banned their employees from working on anything involving GPL 3 code, because of legal uncertainties, especially with regard to patents. Steve From gcd at i.ph Sat Jun 20 17:58:22 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 08:58:22 +0800 Subject: [Freeswitch-users] FS as a Class 5 switch Message-ID: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> hi everybody, i'm interested to know if anyone employed FS as a local exchange switch. i'm confident FS can handle several calls using RTP by-pass mode. however, i'm more concerned on handling the large dialplan with hundreds (or even a few thousand) exchange prefixes nationwide during call setup. i'd be glad to hear experiences and suggestions esp on the hardware dimensioning. we're talking a small exchange up to about 1,100 lines only, mostly linked to the main exchange via MFC-R2. tks, nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/999823bf/attachment.html From mcampbellsmith at gmail.com Sat Jun 20 18:28:27 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 21 Jun 2009 11:28:27 +1000 Subject: [Freeswitch-users] voicemail problem Message-ID: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> Hi! I have a problem with voicemail in that freeswitch fails to let users leave their message. Something wrong in the config I guess. I see this in the logs: 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message] (en:en) 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) I assume the vm-record_message.PCMU is the file that will be created to record the voicemail. Is that correct and how can I fix this? Thanks! From dyfet at gnutelephony.org Sat Jun 20 18:28:10 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Sat, 20 Jun 2009 21:28:10 -0400 Subject: [Freeswitch-users] MPL and licensing In-Reply-To: <4A3D858E.5050203@coppice.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <4A3D858E.5050203@coppice.org> Message-ID: <4A3D8CAA.1040101@gnutelephony.org> There are no legal uncertainties with respect to patents in GPL v3. You cannot assert them in code you license under it. There was ambiguities in GPL v2 in this respect which some companies liked. I prefer to deal with honest companies rather than those that are anti-social or might choose legal ambush later, so any that feel they cannot accept the greater legal certainty of GPL v3 in this respect are probably companies that I would not choose to have any kind of relationship with anyway ;). I recall there were other technical reasons why some have preferred the MPL, especially over the language of the Lesser GNU General Public License prior to v3. I remember having a lovely discussion about this with Craig Southern a few years back who conceeded that if the language (of the older LGPL) had been corrected for C++ use cases and object oriented practices (inlines, templates, derived classes, etc, all were problems...), he would likely have used it at the time instead of the MPL for OpenH323. Steve Underwood wrote: > paul.degt wrote: >> Yes, that's one of the reasons. Another point is that GPL v.3 is defined >> more clearly from legal perspective, at least from our legal adviser >> point of view. >> > While the legal status of MPL is widely considered to be vague, is GPL 3 > any better? GPL 2 is pretty sound, and has stood the test of time. > However a number of large companies have banned their employees from > working on anything involving GPL 3 code, because of legal > uncertainties, especially with regard to patents. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 177 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/ced168ae/attachment.vcf From msc at freeswitch.org Sat Jun 20 18:40:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 20 Jun 2009 20:40:45 -0500 Subject: [Freeswitch-users] FS as a Class 5 switch In-Reply-To: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> References: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> Message-ID: <87f2f3b90906201840h34bce529oa86267ab50c98c08@mail.gmail.com> Isn't Telco Bridges using it as a switch? -MC On Sat, Jun 20, 2009 at 7:58 PM, Nandy Dagondon wrote: > hi everybody, > > i'm interested to know if anyone employed FS as a local exchange switch. > i'm confident FS can handle several calls using RTP by-pass mode. however, > i'm more concerned on handling the large dialplan with hundreds (or even a > few thousand) exchange prefixes nationwide during call setup. > > i'd be glad to hear experiences and suggestions esp on the hardware > dimensioning. we're talking a small exchange up to about 1,100 lines only, > mostly linked to the main exchange via MFC-R2. > > tks, > nandy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/06e46271/attachment.html From steveu at coppice.org Sat Jun 20 18:52:49 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 21 Jun 2009 09:52:49 +0800 Subject: [Freeswitch-users] MPL and licensing In-Reply-To: <4A3D8CAA.1040101@gnutelephony.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <4A3D858E.5050203@coppice.org> <4A3D8CAA.1040101@gnutelephony.org> Message-ID: <4A3D9271.6030903@coppice.org> David Sugar wrote: > There are no legal uncertainties with respect to patents in GPL v3. You > cannot assert them in code you license under it. There was ambiguities > in GPL v2 in this respect which some companies liked. I prefer to deal > with honest companies rather than those that are anti-social or might > choose legal ambush later, so any that feel they cannot accept the > greater legal certainty of GPL v3 in this respect are probably companies > that I would not choose to have any kind of relationship with anyway ;). > If that is true, why are there some notes on the FSF web site trying to clarify what is not well stated about patents in GPL3 itself? You are fooling yourself if you think lawyers are generally comfortable with GPL3. I'm talking about lawyers who were perfectly happy with GPL2. I know of some cases where large companies have paid outsourced developers to contribute to open source projects, specifically so there are no possible legal ramifications related to their own patent portfolio. That's a really messed up licence. Steve From juanma.v82 at gmail.com Sat Jun 20 19:48:39 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Sat, 20 Jun 2009 23:48:39 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: Hi, Saeed thanks for reply my mails. I am only using ONE FS. The Endpoint 1 call to the FS, then it call to SWITCH and the SWITCH call back to the FS and this call to Endpoint 2, my first intention were to use FS in bypass_mode, in this mode Fs only handle SIP signalization and SWITCH (another technology) handle RTP, like the image in the link (http://img26.imageshack.us/img26/1298/fsassbc.png ). Only ONE FS work as SBC (session border controler) or pseudo proxy. But the problem is the FS doesn't negotiate codecs properly and how you wrote late negotiation didn't work for change codec negotiation. I will test your tip, but i cant control the codec negotiation from endpoint who make a first INVITE to FS, only the bridge to de SWICTH and back to FS. Yes, I am using xml_curl. I was testing with event_socket but when FS receive a call and this connect to event_socket, the FS try to park the call and this can't by done because you must control the media. Thanks Regards On 20/06/2009, at 12:06, Saeed Ahmad wrote: > Hi, > > I am not expert, and if i understood you correctly, i am doing it > like that: > > set late negotiation to false (it can be set true if you want to > negotiate it on the fly, but it didn't work for me) > > don't use absolute_codec_string instead use codec_string and put > codec(s) what EP2 is offering, now if A side is sending one of the > codec which are defined for EP2 then it will be used. > > i didn't consider your FS at right side because your SWITCH > definately going to transcode the codec into G729, otherwise above > scenario should be same for both FS. > let me know if it also works for you, i am also using it in proxy > media mode. > are you using xml_curl? > > - Saeed > On Fri, Jun 19, 2009 at 11:28 PM, Mariano de Llano > wrote: > > You are right, it seams that it can not be done. > > In the past I've tried to do something similiar but with no success. > Apperently the documentation is wrong. > > If I have some time I will look at the code and I will give you some > feedback. > > Cheers, > > PS: The transcoding question has nothing to do with your question :) > > On 19/06/2009, at 17:18, JuanMa wrote: > > > I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I > > have another switch who is in charge of it, witch is from another > > technology), I only want to negotiate the codecs in the way that I > > want it. This only seams to work when bypass media or proxy media is > > set to false. Due I need to use it as a SBC(session border > controller) > > or pseudo proxy (I already know that is not intend for it), I need > to > > negotiate the codecs in the FS. In the current thread I've already > > explained what I'm trying to do. If you give me a tip I'm willing to > > make the documentation richer. > > > > Thanks > > Regards > > > > In my architecture the switch who is in charge of transcoding IS > NOT a > > FS. > > > > > > On 19/06/2009, at 16:19, Brian West wrote: > > > >> No right now you can not legally transcode G729 in FreeSWITCH, > >> PERIOD! > >> > >> /b > >> > >> On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > >> > >>> Yes, it can do transcoding. Transcoding isn't the problem to my > >>> architecture, my problem is the codec negotiation between FS and > >>> Endpoints. > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/0d191cbe/attachment-0001.html From dave at 3c.co.uk Sat Jun 20 20:50:04 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 21 Jun 2009 06:50:04 +0300 Subject: [Freeswitch-users] FS as a Class 5 switch In-Reply-To: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> References: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> Message-ID: <1245556204.29626.19.camel@dk-d820> Hi Nandy. On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote: > i'm interested to know if anyone employed FS as a local exchange > switch. i'm confident FS can handle several calls using RTP by-pass > mode. however, i'm more concerned on handling the large dialplan with > hundreds (or even a few thousand) exchange prefixes nationwide during > call setup. We have probably ~100k prefixes in our LCR. We don't put these in the dialplan directly; instead, they live in a database and we have an external application which routes calls. FreeSWITCH has mod_lcr which I would imagine will do the same sort of thing; we don't use it because it wasn't around when we started. I'd caution against trying to put thousands of prefixes in the dialplan: I'd guess that matching each call against some thousands of regexes during call setup might get expensive. > i'd be glad to hear experiences and suggestions esp on the hardware > dimensioning. we're talking a small exchange up to about 1,100 lines > only, mostly linked to the main exchange via MFC-R2. That'd depend on the number of concurrent calls you need to budget for - taking it that 1,100 lines implies maybe 1-200 simultaneous calls, then one low-end modern server (Core 2 Duo, etc.) ought to do just fine. Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From gcd at i.ph Sat Jun 20 21:25:58 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 12:25:58 +0800 Subject: [Freeswitch-users] FS as a Class 5 switch In-Reply-To: <1245556204.29626.19.camel@dk-d820> References: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> <1245556204.29626.19.camel@dk-d820> Message-ID: <7d0bfd8c0906202125uec24ad0tc440437fb1f5134e@mail.gmail.com> hi dave, tks for sharing us this info. i don't think we can reach 10k prefixes but your deployment to use external database or mod_lcr is the way to go. re hardware, i think core2 platform would be enough cuz it will be in a rural installation. i'm sure it wont reach 200 simultaneous calls. FS community is really great! tks once again, nandy On Sun, Jun 21, 2009 at 11:50 AM, David Knell wrote: > Hi Nandy. > > On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote: > > i'm interested to know if anyone employed FS as a local exchange > > switch. i'm confident FS can handle several calls using RTP by-pass > > mode. however, i'm more concerned on handling the large dialplan with > > hundreds (or even a few thousand) exchange prefixes nationwide during > > call setup. > > We have probably ~100k prefixes in our LCR. We don't put these in the > dialplan directly; instead, they live in a database and we have an > external application which routes calls. FreeSWITCH has mod_lcr which I > would imagine will do the same sort of thing; we don't use it because it > wasn't around when we started. > > > I'd caution against trying to put thousands of prefixes in the dialplan: > I'd guess that matching each call against some thousands of regexes > during call setup might get expensive. > > > i'd be glad to hear experiences and suggestions esp on the hardware > > dimensioning. we're talking a small exchange up to about 1,100 lines > > only, mostly linked to the main exchange via MFC-R2. > > That'd depend on the number of concurrent calls you need to budget for - > taking it that 1,100 lines implies maybe 1-200 simultaneous calls, then > one low-end modern server (Core 2 Duo, etc.) ought to do just fine. > Cheers -- > > Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/7c61c594/attachment.html From gcd at i.ph Sat Jun 20 22:36:13 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 13:36:13 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem Message-ID: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> hi, i tested the latest SVN build (13884) using the sample configuration files ... no modifications whatsoever. but in sofia external profile, the IP address is my internal address instead of my external IP address. did i miss something here? tks. -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/099963a6/attachment.html From paul.degt at gmail.com Sat Jun 20 22:38:19 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Sun, 21 Jun 2009 01:38:19 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D858E.5050203@coppice.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <4A3D858E.5050203@coppice.org> Message-ID: <4A3DC74B.4080404@gmail.com> I have Fortune 1000 clients myself, and frankly speaking in real world they don't even care what type of license the free stuff has. Why? Simple. Because 90% of the time these companies buy commercial support and being a commercial customer it's very easy for them to get a commercial version of the software for a reasonable price. But the latter not happening very often - most of the time they never publish their in-house software since it's their competitive advantage - and thus no reason to be concerned about license type. As to "legal uncertainties" - frankly I don't know, my lawyer liked it, knowing how complex the law can be I am sure any type of license free or non-free can potentially bring legal issues - it's imperfect world. Steve Underwood wrote: > paul.degt wrote: > >> Yes, that's one of the reasons. Another point is that GPL v.3 is defined >> more clearly from legal perspective, at least from our legal adviser >> point of view. >> >> > While the legal status of MPL is widely considered to be vague, is GPL 3 > any better? GPL 2 is pretty sound, and has stood the test of time. > However a number of large companies have banned their employees from > working on anything involving GPL 3 code, because of legal > uncertainties, especially with regard to patents. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jason at jasonjgw.net Sat Jun 20 22:45:54 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 21 Jun 2009 15:45:54 +1000 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> Message-ID: <20090621054554.GA24069@jdc.jasonjgw.net> Nandy Dagondon wrote: > hi, > > i tested the latest SVN build (13884) using the sample configuration files > ... no modifications whatsoever. but in sofia external profile, the IP > address is my internal address instead of my external IP address. > > did i miss something here? Try setting ext-sip-ip and ext-rtp-ip in the external profile to stun:stun.freeswitch.org This can alternatively be set using global variables in vars.xml in the supplied configuration. From gcd at i.ph Sat Jun 20 23:20:10 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 14:20:10 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <20090621054554.GA24069@jdc.jasonjgw.net> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> Message-ID: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> the default setting is "auto-nat". i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun:stun.freeswitch.org. result: same problem i tried your suggestion. still the same problem. On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: > Nandy Dagondon wrote: > > hi, > > > > i tested the latest SVN build (13884) using the sample configuration > files > > ... no modifications whatsoever. but in sofia external profile, the IP > > address is my internal address instead of my external IP address. > > > > did i miss something here? > > Try setting ext-sip-ip and ext-rtp-ip in the external profile to > stun:stun.freeswitch.org > > This can alternatively be set using global variables in vars.xml in the > supplied configuration. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/4f2de991/attachment.html From jmesquita at gmail.com Sat Jun 20 23:35:53 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 21 Jun 2009 03:35:53 -0300 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> Message-ID: <5a8712120906202335h1e7a18efo9791a901c6b83c2f@mail.gmail.com> What I would guess is the the STUN lookup failed. Do you have anything on this box that would prevent FS from doing DNS lookup? jmesquita On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and > ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun: > stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. > > > > On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: > >> Nandy Dagondon wrote: >> > hi, >> > >> > i tested the latest SVN build (13884) using the sample configuration >> files >> > ... no modifications whatsoever. but in sofia external profile, the IP >> > address is my internal address instead of my external IP address. >> > >> > did i miss something here? >> >> Try setting ext-sip-ip and ext-rtp-ip in the external profile to >> stun:stun.freeswitch.org >> >> This can alternatively be set using global variables in vars.xml in the >> supplied configuration. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/00aefa7d/attachment-0001.html From gcd at i.ph Sat Jun 20 23:55:08 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 14:55:08 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <5a8712120906202335h1e7a18efo9791a901c6b83c2f@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> <5a8712120906202335h1e7a18efo9791a901c6b83c2f@mail.gmail.com> Message-ID: <7d0bfd8c0906202355r70e57008i78c9ff1ad9df9020@mail.gmail.com> i just come across the Auto NAT feature in the Wiki. i'm testing if my router UPNP works w/ FS. STUN works before i updated to SVN. 2009/6/21 Jo?o Mesquita > What I would guess is the the STUN lookup failed. Do you have anything on > this box that would prevent FS from doing DNS lookup? > > jmesquita > > > On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon wrote: > >> the default setting is "auto-nat". >> >> i changed ext-sip-ip=$${external_sip_ip} and >> ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun: >> stun.freeswitch.org. result: same problem >> >> i tried your suggestion. still the same problem. >> >> >> >> On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: >> >>> Nandy Dagondon wrote: >>> > hi, >>> > >>> > i tested the latest SVN build (13884) using the sample configuration >>> files >>> > ... no modifications whatsoever. but in sofia external profile, the IP >>> > address is my internal address instead of my external IP address. >>> > >>> > did i miss something here? >>> >>> Try setting ext-sip-ip and ext-rtp-ip in the external profile to >>> stun:stun.freeswitch.org >>> >>> This can alternatively be set using global variables in vars.xml in the >>> supplied configuration. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/5e68ab83/attachment.html From gcd at i.ph Sun Jun 21 00:38:25 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 15:38:25 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> Message-ID: <7d0bfd8c0906210038o17798500s2ae50d6bd9a43f80@mail.gmail.com> it's working now, i mean the Auto NAT feature - after i enabled UPNP feature on my router. it's based on external IP addresses of Ext-SIP-IP and Ext-RTP-IP when performing "sofia status profile [internal|external]" on the cli. however, "sofia status" still shows internal IP address on the external profile. it should display the external IP address instead. On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and > ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun: > stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. > > > > On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: > >> Nandy Dagondon wrote: >> > hi, >> > >> > i tested the latest SVN build (13884) using the sample configuration >> files >> > ... no modifications whatsoever. but in sofia external profile, the IP >> > address is my internal address instead of my external IP address. >> > >> > did i miss something here? >> >> Try setting ext-sip-ip and ext-rtp-ip in the external profile to >> stun:stun.freeswitch.org >> >> This can alternatively be set using global variables in vars.xml in the >> supplied configuration. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/0fa7574f/attachment.html From mike at jerris.com Sun Jun 21 01:27:36 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 Jun 2009 04:27:36 -0400 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906210038o17798500s2ae50d6bd9a43f80@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> <7d0bfd8c0906210038o17798500s2ae50d6bd9a43f80@mail.gmail.com> Message-ID: <06F7DAC3-2586-4422-825C-93E1CAE43D0C@jerris.com> It has both the interface bind address and the external address Mike On Jun 21, 2009, at 3:38 AM, Nandy Dagondon wrote: > it's working now, i mean the Auto NAT feature - after i enabled UPNP > feature on my router. it's based on external IP addresses of Ext-SIP- > IP and Ext-RTP-IP when performing "sofia status profile [internal| > external]" on the cli. > > however, "sofia status" still shows internal IP address on the > external profile. it should display the external IP address instead. > > > On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$$ > {external_rtp_ip}. both of them are set in vars.xml as > stun:stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. > > > > On Sun, Jun 21, 2009 at 1:45 PM, Jason White > wrote: > Nandy Dagondon wrote: > > hi, > > > > i tested the latest SVN build (13884) using the sample > configuration files > > ... no modifications whatsoever. but in sofia external profile, > the IP > > address is my internal address instead of my external IP address. > > > > did i miss something here? > > Try setting ext-sip-ip and ext-rtp-ip in the external profile to > stun:stun.freeswitch.org > > This can alternatively be set using global variables in vars.xml in > the > supplied configuration. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/619bdc3f/attachment.html From christian.loeschenkohl at xpirio.com Sun Jun 21 03:05:59 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sun, 21 Jun 2009 12:05:59 +0200 Subject: [Freeswitch-users] channel variable sip_to_tag Message-ID: <4A3E0607.1060608@xpirio.com> hello do someone know how to get the sip_to_tag from an active call? the sip_from_tag is available as a channel variable but sip_to_tag isn't. i don't know if it is available at call setup, the fist time i see the tag=... in the sip header is the challenge response answer from fs i need this to get my aoc (advice-of-charge) implementation running, this one is based on sip info messages and has to contain the same tag's as the active call. br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From brian at freeswitch.org Sun Jun 21 03:12:03 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 05:12:03 -0500 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> Message-ID: <617EB39F-ED75-4969-B79F-4DD7D20E6364@freeswitch.org> Really it shouldn't have changed unless you wiped your configs. The reason it can't work with auto-nat is you're not behind a natpmp or upnp router thus you're going to have to set them manually use stun. You can not use stun for rtp-ip or sip-ip, just for ext-sip-ip and ext- rtp-ip once they are set correctly it should work without a problem. /b On Jun 21, 2009, at 1:20 AM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$$ > {external_rtp_ip}. both of them are set in vars.xml as > stun:stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/884ae856/attachment-0001.html From jan.kubr at gmail.com Sun Jun 21 04:04:25 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 13:04:25 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT Message-ID: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> We have a SIP gateway behind NAT which I haven't been able to set up to work with Freeswitch. The configuration I thought would work is: This, however, sends REGISTER sip:78.24.13.197;transport=udp SIP/2.0 whereas I need REGISTER sip:11.12.13.10;transport=udp SIP/2.0 When I set "register-proxy" to the public address (78.24.13.197) and comment the "proxy" param out, it successfully registers. Though any other SIP messages are sent to 11.12.13.10 which obviously fails. Basically what I need Freeswitch to do is to send everything to the public address, but put the private one everywhere in the SIP messages. How do I do this? Thanks, Jan From mcampbellsmith at gmail.com Sun Jun 21 04:27:58 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 21 Jun 2009 21:27:58 +1000 Subject: [Freeswitch-users] email core dump Message-ID: <33c87fa30906210427x6833e0adk46d313247ad00ef5@mail.gmail.com> Hi! I am trying to email from 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to 1000 at 192.168.0.20 /bin/cat: write error: Broken pipe sh: line 1: 11975 Done(1) /bin/cat /tmp/mail.124558382500b1 11976 Segmentation fault (core dumped) | exim4 -t myemail at xx.com 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.12455810042c7f] to [myemail at xx.com] I can manually send an email to myself with exim4, but freeswitch fails. Any ideas what I have configured incorrectly? Thanks From jan.kubr at gmail.com Sun Jun 21 04:39:26 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 13:39:26 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> Message-ID: <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> I have found this: http://jira.freeswitch.org/browse/MODENDP-184Thanks to which I know that adding to the profile XML does almost what I need. Is it possible to configure outbound proxy per gateway though? Cheers, Jan On Sun, Jun 21, 2009 at 1:04 PM, Jan Kubr wrote: > We have a SIP gateway behind NAT which I haven't been able to set up > to work with Freeswitch. The configuration I thought would work is: > > > > > > > > > > > This, however, sends > > REGISTER sip:78.24.13.197;transport=udp SIP/2.0 > > whereas I need > > REGISTER sip:11.12.13.10;transport=udp SIP/2.0 > > When I set "register-proxy" to the public address (78.24.13.197) and > comment the "proxy" param out, it successfully registers. Though any > other SIP messages are sent to 11.12.13.10 which obviously fails. > > Basically what I need Freeswitch to do is to send everything to the > public address, but put the private one everywhere in the SIP > messages. How do I do this? > > Thanks, > Jan > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/d7337be1/attachment.html From jan.kubr at gmail.com Sun Jun 21 05:16:11 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 14:16:11 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> Message-ID: <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> Creating a separate sofia profile just for this gateway definitely works, just wondering whether there is a cleaner solution. The register-proxy params seems to do something very similar.. On Sun, Jun 21, 2009 at 1:39 PM, Jan Kubr wrote: > I have found this: http://jira.freeswitch.org/browse/MODENDP-184 Thanks to > which I know that adding > > > > to the profile XML does almost what I need. Is it possible to configure > outbound proxy per gateway though? > > Cheers, > Jan > > On Sun, Jun 21, 2009 at 1:04 PM, Jan Kubr wrote: > >> We have a SIP gateway behind NAT which I haven't been able to set up >> to work with Freeswitch. The configuration I thought would work is: >> >> >> >> >> >> >> >> >> >> >> This, however, sends >> >> REGISTER sip:78.24.13.197;transport=udp SIP/2.0 >> >> whereas I need >> >> REGISTER sip:11.12.13.10;transport=udp SIP/2.0 >> >> When I set "register-proxy" to the public address (78.24.13.197) and >> comment the "proxy" param out, it successfully registers. Though any >> other SIP messages are sent to 11.12.13.10 which obviously fails. >> >> Basically what I need Freeswitch to do is to send everything to the >> public address, but put the private one everywhere in the SIP >> messages. How do I do this? >> >> Thanks, >> Jan >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/dfeea9c2/attachment.html From brian at freeswitch.org Sun Jun 21 05:17:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 07:17:21 -0500 Subject: [Freeswitch-users] email core dump In-Reply-To: <33c87fa30906210427x6833e0adk46d313247ad00ef5@mail.gmail.com> References: <33c87fa30906210427x6833e0adk46d313247ad00ef5@mail.gmail.com> Message-ID: <7C7A8ED9-ECED-4100-87F6-0875C054EC64@freeswitch.org> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings /b On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: > Hi! > > I am trying to email from > 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore > original codec. > 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to > 1000 at 192.168.0.20 > /bin/cat: write error: Broken pipe > sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. > 124558382500b1 > 11976 Segmentation fault (core dumped) | exim4 -t myemail at xx.com > 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file > [/tmp/mail.12455810042c7f] to [myemail at xx.com] > > I can manually send an email to myself with exim4, but freeswitch > fails. > > Any ideas what I have configured incorrectly? > > Thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sun Jun 21 05:19:22 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 07:19:22 -0500 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> Message-ID: They usually will not auth on the RURI... I recommend you set the from- domain on your gateway... I think thats really what you need. /b On Jun 21, 2009, at 6:39 AM, Jan Kubr wrote: > I have found this: http://jira.freeswitch.org/browse/MODENDP-184 > Thanks to which I know that adding > > > > to the profile XML does almost what I need. Is it possible to > configure outbound proxy per gateway though? > > Cheers, > Jan > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/4050689d/attachment.html From brian at freeswitch.org Sun Jun 21 05:25:08 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 07:25:08 -0500 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> Message-ID: <981C2E3E-3BF6-4629-A0EE-FFB47B792104@freeswitch.org> nobody authenticates on the request URI... you're focusing on the wrong thing... you'll need from-domain and/or from-user I suspect. /b On Jun 21, 2009, at 7:16 AM, Jan Kubr wrote: > Creating a separate sofia profile just for this gateway definitely > works, just wondering whether there is a cleaner solution. The > register-proxy params seems to do something very similar.. From larclap at yahoo.com Sun Jun 21 10:36:19 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sun, 21 Jun 2009 10:36:19 -0700 Subject: [Freeswitch-users] svn update error Message-ID: <006f01c9f296$c99f0d30$5cdd2790$@com> I am currently running 13723 and want to get current. When I issue the "svn update" command, the following error appears: A src/mod/asr_tts/mod_unimrcp A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.2008.vcproj A src/mod/asr_tts/mod_unimrcp/unimrcp.vsprops A src/mod/asr_tts/mod_unimrcp/Makefile.am A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c U src/mod/asr_tts/mod_pocketsphinx/mod_pocketsphinx.c U src/mod/event_handlers/mod_event_multicast/mod_event_multicast.c svn: Failed to add file 'src/mod/event_handlers/mod_event_multicast/Makefile': an unversioned file of the same name already exists What do I need to do? Should I delete the Makefile file? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/3f8a1a89/attachment-0001.html From mrene_lists at avgs.ca Sun Jun 21 10:38:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 21 Jun 2009 13:38:17 -0400 Subject: [Freeswitch-users] svn update error In-Reply-To: <006f01c9f296$c99f0d30$5cdd2790$@com> References: <006f01c9f296$c99f0d30$5cdd2790$@com> Message-ID: yeah Math On 21-Jun-09, at 1:36 PM, Lars Zeb wrote: > I am currently running 13723 and want to get current. When I issue > the ?svn update? command, the following error appears: > > A src/mod/asr_tts/mod_unimrcp > A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.2008.vcproj > A src/mod/asr_tts/mod_unimrcp/unimrcp.vsprops > A src/mod/asr_tts/mod_unimrcp/Makefile.am > A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c > U src/mod/asr_tts/mod_pocketsphinx/mod_pocketsphinx.c > U src/mod/event_handlers/mod_event_multicast/mod_event_multicast.c > svn: Failed to add file 'src/mod/event_handlers/mod_event_multicast/ > Makefile': an unversioned file of the same name already exists > > What do I need to do? Should I delete the Makefile file? > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/163338ed/attachment.html From jan.kubr at gmail.com Sun Jun 21 10:44:46 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 19:44:46 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <981C2E3E-3BF6-4629-A0EE-FFB47B792104@freeswitch.org> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> <981C2E3E-3BF6-4629-A0EE-FFB47B792104@freeswitch.org> Message-ID: <698401620906211044u7f989321k769481e1bf3943c7@mail.gmail.com> Well when I duplicate the external profile, add to it and make the gateway configuration this: Then it does what I need. I was just curious if I can achieve the same without creating a new profile. Jan On Sun, Jun 21, 2009 at 2:25 PM, Brian West wrote: > nobody authenticates on the request URI... you're focusing on the > wrong thing... you'll need from-domain and/or from-user I suspect. > > /b > > On Jun 21, 2009, at 7:16 AM, Jan Kubr wrote: > > > Creating a separate sofia profile just for this gateway definitely > > works, just wondering whether there is a cleaner solution. The > > register-proxy params seems to do something very similar.. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/b10f3398/attachment.html From william.suffill at gmail.com Sun Jun 21 11:44:15 2009 From: william.suffill at gmail.com (William Suffill) Date: Sun, 21 Jun 2009 14:44:15 -0400 Subject: [Freeswitch-users] svn update error In-Reply-To: References: <006f01c9f296$c99f0d30$5cdd2790$@com> Message-ID: <6b65470d0906211144l1bf9f6feyaee0854fad28eece@mail.gmail.com> Seems there was some major changes depending on what revision you are trying to update from. A fresh SVN checkout might be the fastest fix and what I did here once I couldn't update very old checkouts. Technically if you slowly updated it from revision -> revision until it was current that too should work. Not worth the hassle IMHO tho. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/e23693b4/attachment.html From drago at windstream.net Sun Jun 21 12:51:31 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 21 Jun 2009 15:51:31 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: <000001c9ed45$72058320$56108960$@net> References: <000001c9ed45$72058320$56108960$@net> Message-ID: <00cd01c9f2a9$ad1ec700$075c5500$@net> After days of running around clueless, I have no other option but to ask the community for help one last time. Here is what happens: 1. FS sends INVITE 2. Exchange answers with "302 Moved Temporarily" 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: "MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The 'From' field's address to be in the format ''; otherwise, MS Exchange drops the call." I don't know if this is the only problem. However, I see exactly this behavior: "PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established." After "302 (Moved Temporarily", FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/3dbb2445/attachment-0001.html From mike at jerris.com Sun Jun 21 13:31:04 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 Jun 2009 16:31:04 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: <00cd01c9f2a9$ad1ec700$075c5500$@net> References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: > After days of running around clueless, I have no other option but to > ask the community for help one last time? > > > > Here is what happens: > > > > 1. FS sends INVITE > > 2. Exchange answers with ?302 Moved Temporarily? > > 3. FS bombs and closes > > > > > > Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/ > 2.0 > > Message Header > > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > Max-Forwards: 68 > > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > To: > > SIP to address: sip:4783874764 at 10.0.0.71 > > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > CSeq: 116688296 INVITE > > Contact: > > Contact Binding: +um.gmc.cc.ga.us at 10.8.4.3:5060;transport=tcp> > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Privacy: none > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 236 > > P-Asserted-Identity: "MILLEDGEVL GA" > > > Message Body > > > > > > > > Status-Line: SIP/2.0 100 Trying > > Message Header > > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > TO: > > SIP to address: sip:4783874764 at 10.0.0.71 > > CSEQ: 116688296 INVITE > > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > CONTENT-LENGTH: 0 > > > > > > Status-Line: SIP/2.0 302 Moved Temporarily > > Message Header > > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > TO: ;tag=27df6afe0 > > SIP to address: sip:4783874764 at 10.0.0.71 > > SIP tag: 27df6afe0 > > CSEQ: 116688296 INVITE > > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > CONTACT: > > Contact Binding: 4783874764 at 10.0.0.71:5065;transport=tcp> > > CONTENT-LENGTH: 0 > > SERVER: RTCC/3.0.0.0 > > > > > > Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 > > Message Header > > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > Max-Forwards: 68 > > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > To: ;tag=27df6afe0 > > SIP to address: sip:4783874764 at 10.0.0.71 > > SIP tag: 27df6afe0 > > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > CSeq: 116688296 ACK > > Content-Length: 0 > > > > > > Here is the FS log beginning the the processing of the call: > > > > 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel > sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937- > b321c0d87414] > > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/ > external/4782512197 at 209.249.3.59 entering state [received][100] > > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=NXT02 19785 8060 IN IP4 209.249.3.59 > > s=sip call > > c=IN IP4 209.249.3.60 > > t=0 0 > > m=audio 36292 RTP/AVP 0 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec > Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/ > external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples > > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf > payload to 101 > > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW > > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 > (sofia/external/4782512197 at 209.249.3.59) State NEW > > 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_NEW -> CS_INIT > > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 > (sofia/external/4782512197 at 209.249.3.59) State INIT > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 > SOFIA INIT > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_INIT -> CS_ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 > (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change > CS_ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 > (sofia/external/4782512197 at 209.249.3.59) State ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 > SOFIA ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 > sofia/external/4782512197 at 209.249.3.59 Standard ROUTING > > 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing > MILLEDGEVL GA->4783874190 in context public > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >unloop] continue=false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >outside_call] continue=true > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition > [outside_call] > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > set(outside_call=true) > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >call_debug] continue=true > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [call_debug] ${call_debug}(true) =~ /^true$/ break=never > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >public_extensions] continue=false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9]) > $/ break=on-false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >Local_UM] continue=false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on- > false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 > (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> > CS_EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 > (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change > CS_EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 > (sofia/external/4782512197 at 209.249.3.59) State EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 > SOFIA EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 > sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE > > EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) > > 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 > SET [outside_call]=[true] > > EXECUTE sofia/external/4782512197 at 209.249.3.59 info() > > 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: > > Channel-State: [CS_EXECUTE] > > Channel-State-Number: [4] > > Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > > Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > > Call-Direction: [inbound] > > Presence-Call-Direction: [inbound] > > Answer-State: [ringing] > > Channel-Read-Codec-Name: [PCMU] > > Channel-Read-Codec-Rate: [8000] > > Channel-Write-Codec-Name: [PCMU] > > Channel-Write-Codec-Rate: [8000] > > Caller-Username: [4782512197] > > Caller-Dialplan: [XML] > > Caller-Caller-ID-Name: [MILLEDGEVL GA] > > Caller-Caller-ID-Number: [4782512197] > > Caller-Network-Addr: [209.249.3.59] > > Caller-Destination-Number: [4783874190] > > Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > > Caller-Source: [mod_sofia] > > Caller-Context: [public] > > Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > > Caller-Profile-Index: [1] > > Caller-Profile-Created-Time: [1245613758636018] > > Caller-Channel-Created-Time: [1245613758636018] > > Caller-Channel-Answered-Time: [0] > > Caller-Channel-Progress-Time: [0] > > Caller-Channel-Progress-Media-Time: [0] > > Caller-Channel-Hangup-Time: [0] > > Caller-Channel-Transfer-Time: [0] > > Caller-Screen-Bit: [true] > > Caller-Privacy-Hide-Name: [false] > > Caller-Privacy-Hide-Number: [false] > > variable_sip_received_ip: [209.249.3.59] > > variable_sip_received_port: [5060] > > variable_sip_via_protocol: [udp] > > variable_sip_from_user: [4782512197] > > variable_sip_from_uri: [4782512197 at 209.249.3.59] > > variable_sip_from_host: [209.249.3.59] > > variable_sip_from_user_stripped: [4782512197] > > variable_sip_from_tag: [3454602382-411732] > > variable_sofia_profile_name: [external] > > variable_sip_cid_type: [pid] > > variable_sip_req_user: [gw+Broadvox] > > variable_sip_req_port: [5080] > > variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] > > variable_sip_req_host: [71.29.0.61] > > variable_sip_to_user: [4783874764] > > variable_sip_to_port: [5060] > > variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] > > variable_sip_to_host: [209.249.3.56] > > variable_sip_contact_user: [4782512197] > > variable_sip_contact_port: [5060] > > variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] > > variable_sip_contact_host: [209.249.3.59] > > variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] > > variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] > > variable_sip_via_host: [209.249.3.59] > > variable_sip_via_port: [5060] > > variable_max_forwards: [69] > > variable_sip_call_info: [ 209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"] > > variable_sip_gateway: [Broadvox] > > variable_switch_r_sdp: [v=0 > > o=NXT02 19785 8060 IN IP4 209.249.3.59 > > s=sip call > > c=IN IP4 209.249.3.60 > > t=0 0 > > m=audio 36292 RTP/AVP 0 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > ] > > variable_remote_media_ip: [209.249.3.60] > > variable_remote_media_port: [36292] > > variable_read_codec: [PCMU] > > variable_read_rate: [8000] > > variable_write_codec: [PCMU] > > variable_write_rate: [8000] > > variable_endpoint_disposition: [RECEIVED] > > variable_outside_call: [true] > > variable_current_application: [info] > > > > > > EXECUTE sofia/external/4782512197 at 209.249.3.59 > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > > 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 > variable string 0 = [absolute_codec_string=PCMA] > > 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] > > 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/ > 4783874764) State Change CS_NEW -> CS_INIT > > 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_INIT > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT > > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 > SOFIA INIT > > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 4783874764) State Change CS_INIT -> CS_ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT going to sleep > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 > SOFIA ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/ > internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING going to sleep > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 > (sofia/internal/4783874764) State CONSUME_MEDIA > > 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/ > internal/4783874764 entering state [calling][0] > > > > > > Will trade my first born for little help J > > > > Drago > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Drago Totev > Sent: Sunday, June 14, 2009 7:11 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 > > > > Hello everyone, > > > > I am trying to set a test environment with FS and Exchange 2007. > > > > Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM > and it does not seem to work. Thus far I use the default settings, > one ATA registered and confirmed to work. > > > > According external source: > > > > ?MS Exchange does not understand 'From' fields with domain name in t > he address (example: field containing addre > ss).The ?From? field?s address to be in the format > ??; otherwise, MS Exchange drops the call.? > > > > I don?t know if this is the only problem? However, I see exactly thi > s behavior: > > > > ?PBX initiates call to MS Exchange by sending a SIP INVITE message t > o the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Tempora > rily) response to PBX asking to repeat INVITE to a different port (5 > 065 for example). After PBX repeats the INVITE sending, the call is > established.? > > > > After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without > any record in the log. > > > > Can someone help with working configuration, please? Exchange 2007 > UM role Version: 08.01.0359.002 > > > > Thanks in advance. > > > > Drago > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/85be5cc9/attachment-0001.html From drago at windstream.net Sun Jun 21 14:21:07 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 21 Jun 2009 17:21:07 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: <00de01c9f2b6$31d14de0$9573e9a0$@net> Wiki does not have instruction how to do so on Windows (I should have mention this in the beginning ? sorry about that). Can I do it anyway? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Sunday, June 21, 2009 4:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: After days of running around clueless, I have no other option but to ask the community for help one last time? Here is what happens: 1. FS sends INVITE 2. Exchange answers with ?302 Moved Temporarily? 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: ?MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The ?From? field?s address to be in the format ??; otherwise, MS Exchange drops the call.? I don?t know if this is the only problem? However, I see exactly this behavior: ?PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established.? After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/3c4be4b6/attachment-0001.html From drago at windstream.net Sun Jun 21 14:23:45 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 21 Jun 2009 17:23:45 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: <00e301c9f2b6$8f710c10$ae532430$@net> The Windows Application Log shows this, however: 1331177817 1 APPCRASH Not available 0 FreeSwitch.exe 0.0.0.0 4a316e86 MSVCR90.dll 9.0.30729.4918 49d43da7 c0000005 0003b590 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Sunday, June 21, 2009 4:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: After days of running around clueless, I have no other option but to ask the community for help one last time? Here is what happens: 1. FS sends INVITE 2. Exchange answers with ?302 Moved Temporarily? 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: ?MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The ?From? field?s address to be in the format ??; otherwise, MS Exchange drops the call.? I don?t know if this is the only problem? However, I see exactly this behavior: ?PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established.? After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/ea5968f1/attachment-0001.html From larclap at yahoo.com Sun Jun 21 17:42:40 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sun, 21 Jun 2009 17:42:40 -0700 Subject: [Freeswitch-users] event_add_head warning message on cosole Message-ID: <00bf01c9f2d2$594f1a20$0bed4e60$@com> I just upgraded to version 13886. On the console the following messages appear every few minutes. I've looked at the code but it's way over my head. Why is it displaying? How can I turn it off? 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'toll_allow' = 'domestic,international,local' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'accountcode' = '1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'user_context' = 'default' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'effective_caller_id_name' = 'Extension 1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'effective_caller_id_number' = '1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'outbound_caller_id_number' = '3235551212' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'callgroup' = 'techsupport' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'record_stereo' = 'true' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'default_gateway' = 'example.com' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'default_areacode' = '323' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'transfer_fallback_extension' = 'operator' Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/52305a54/attachment.html From JCasale at activenetwerx.com Sun Jun 21 18:25:56 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 01:25:56 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk Message-ID: I attempted to build rpm's from the included spec file using a non-root user build environment. The steps I used are as follows: 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as expected by the spec: svn co http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4 tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ 3. Copy spec to SPECS directory: 4. Pull in Source (Source0 doesn't exist yet, I make it above): for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done 5. Check spec in svn copy for deps: yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' freeswitch.spec) 6.Build rpm: rpmbuild -ba freeswitch.spec After some time, near the end I see various issues like the following: making install mod_speex installing mod_speex.so quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/freeswitch-1.0.4/libfreeswitch.la' has not been installed in `/opt/freeswitch/lib' It also seems to download everything it would normally again, then fails with several errors like the following: RPM build errors: File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/opt/freeswitch/mod/ozmod_wanpipe.so* Installed (but unpackaged) file(s) found: /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml . . . After which no rpm's are built? Anyone know what tricks are still needed with the spec from svn? Thanks! jlc From mike at jerris.com Sun Jun 21 18:45:01 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 Jun 2009 21:45:01 -0400 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: Message-ID: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> It's failing to build the core library. There should be some warning before it tried to build the modules in the log. On Jun 21, 2009, at 9:25 PM, "Joseph L. Casale" wrote: > I attempted to build rpm's from the included spec file using a non- > root user > build environment. The steps I used are as follows: > > 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS > 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as > expected by the spec: > svn co http://svn.freeswitch.org/svn/freeswitch/trunk > freeswitch-1.0.4 > tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ > 3. Copy spec to SPECS directory: > 4. Pull in Source (Source0 doesn't exist yet, I make it above): > for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' > freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done > 5. Check spec in svn copy for deps: > yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' > freeswitch.spec) > 6.Build rpm: > rpmbuild -ba freeswitch.spec > > After some time, near the end I see various issues like the following: > > making install mod_speex > installing mod_speex.so > quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/ > freeswitch-1.0.4/libfreeswitch.la' > has not been installed in `/opt/freeswitch/lib' > > > It also seems to download everything it would normally again, then > fails with > several errors like the following: > > RPM build errors: > File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/ > opt/freeswitch/mod/ozmod_wanpipe.so* > Installed (but unpackaged) file(s) found: > /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml > /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml > /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml > /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml > . > . > . > > After which no rpm's are built? Anyone know what tricks are still > needed with > the spec from svn? > > Thanks! > jlc > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Sun Jun 21 21:08:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 22 Jun 2009 00:08:07 -0400 Subject: [Freeswitch-users] event_add_head warning message on cosole In-Reply-To: <00bf01c9f2d2$594f1a20$0bed4e60$@com> References: <00bf01c9f2d2$594f1a20$0bed4e60$@com> Message-ID: <802DCA59-126E-4D33-A910-C95C7644CACB@avgs.ca> 13887 created by brian on 21 June 2009, 21:52:31 -0500 (75 minutes ago) (patch) move this to debug and profile->debug so that its not on unless you enable the profile debug also. That should fix it, update again :) Math On 21-Jun-09, at 8:42 PM, Lars Zeb wrote: > I just upgraded to version 13886. On the console the following > messages appear every few minutes. I?ve looked at the code but it?s > way over my head. > > Why is it displaying? How can I turn it off? > > 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 > event_add_header -> 'toll_allow' = 'domestic,international,local' > 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 > event_add_header -> 'accountcode' = '1000' > 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 > event_add_header -> 'user_context' = 'default' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/61fa3c57/attachment.html From brian at freeswitch.org Sun Jun 21 22:51:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 01:51:02 -0400 Subject: [Freeswitch-users] event_add_head warning message on cosole In-Reply-To: <802DCA59-126E-4D33-A910-C95C7644CACB@avgs.ca> References: <00bf01c9f2d2$594f1a20$0bed4e60$@com> <802DCA59-126E-4D33-A910-C95C7644CACB@avgs.ca> Message-ID: <07B4A4D4-06E4-4A88-8E96-204C971D246A@freeswitch.org> it was actually tony but who cares! ;) /b On Jun 22, 2009, at 12:08 AM, Mathieu Rene wrote: > 13887 created by brian on 21 June 2009, 21:52:31 -0500 (75 minutes > ago) (patch) move this to debug and profile->debug so that its not > on unless you enable the profile debug also. > > That should fix it, update again :) > > Math > > On 21-Jun-09, at 8:42 PM, Lars Zeb wrote: > >> I just upgraded to version 13886. On the console the following >> messages appear every few minutes. I?ve looked at the code but it?s >> way over my head. >> >> Why is it displaying? How can I turn it off? >> >> 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 >> event_add_header -> 'toll_allow' = 'domestic,international,local' >> 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 >> event_add_header -> 'accountcode' = '1000' >> 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 >> event_add_header -> 'user_context' = 'default' > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b8bdebec/attachment-0001.html From d at d-man.org Mon Jun 22 00:31:57 2009 From: d at d-man.org (Darren Schreiber) Date: Mon, 22 Jun 2009 00:31:57 -0700 Subject: [Freeswitch-users] Question about bridging calls to a specific URI via a specific profile Message-ID: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> Hello, I was wondering, I am bridging a call to a specific URI as follows: EXECUTE sofia/internal/+17209460000@ 2.3.4.5 bridge(sofia/internal/3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip at 3As%403.55.66.180%3A7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/URI at domain that FS will use the specified SIP profile to try and connect a call to the URI at domain specified. Since the full URI at domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:XXX at domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/5b6f5687/attachment.html From d at d-man.org Mon Jun 22 01:06:16 2009 From: d at d-man.org (Darren Schreiber) Date: Mon, 22 Jun 2009 01:06:16 -0700 Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile In-Reply-To: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> References: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> Message-ID: <6EA6ABD53A184F1F90CF51F883729E8E@test> Ignore this thread. Apparently I was stripping sip: from the prefix. I guess you have to specify sip: before utilizing fs_nat and fs_path variables. My bad. _____ From: Darren Schreiber [mailto:d at d-man.org] Sent: Monday, June 22, 2009 12:32 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile Hello, I was wondering, I am bridging a call to a specific URI as follows: EXECUTE sofia/internal/+17209460000@ 2.3.4.5 bridge(sofia/internal/3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip at 3As%403.55.66.180%3A7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/URI at domain that FS will use the specified SIP profile to try and connect a call to the URI at domain specified. Since the full URI at domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:XXX at domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b78961e6/attachment.html From dome at tel.co.th Mon Jun 22 01:10:28 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 22 Jun 2009 15:10:28 +0700 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> Message-ID: <8ccbff060906220110y2b9431b5g797d4122152891d9@mail.gmail.com> default_language still don't work wirh say but sound_prefix work fine. example my javascript ------------------------------- session.execute("set", "sound_prefix=/opt/freeswitch/sounds/th"); session.execute("say","th number pronounced 1346523"); session.execute("say","th number pronounced 21"); session.execute("say","th number pronounced 11"); session.execute("say","th number pronounced 101"); How to check in mod_say_th back to freeswotch ? Dome C. 2009/6/3 Brian West : > You'll need to set the variable?default_language > /b > On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote: > > Dear sir, > ????????i create mod_say_th for Thai language. i found some problem > about sound-path. > I have config th.xml in conf/lang/th/ > tts-engine="cepstral" tts-voice="callie"> > ... > > when i try > > Freeswitch still looking sounf file ?in ?/sounds/en/us/callie ?(en > sound-path) > > Someone help me please > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Mon Jun 22 01:38:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 01:38:41 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED Message-ID: <24143545.post@talk.nabble.com> Hi, API CALL [originate sofia/external/1001 at 116.50.456.212] -ERR SERVICE_NOT_IMPLEMENTED I receiving this error i dont know y? Can u help mo on this? I dialing a mobile number on this sometimes it works... Sometimes it destroys the call [CALL_DESTROY] Thanks -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24143545.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 22 01:39:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 01:39:59 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED Message-ID: <24143545.post@talk.nabble.com> Hi, API CALL [originate sofia/external/1001 at 116.50.456.212 1001] output: -ERR SERVICE_NOT_IMPLEMENTED I receiving this error i dont know y? Can u help mo on this? I dialing a mobile number on this sometimes it works... Sometimes it destroys the call [CALL_DESTROY] I already installed zrtp softphone connecting to mobile phones... Thanks -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24143545.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From maarten at nalawo.com Mon Jun 22 02:49:09 2009 From: maarten at nalawo.com (MaartenDM) Date: Mon, 22 Jun 2009 02:49:09 -0700 (PDT) Subject: [Freeswitch-users] Nibblebill heartbeat on B-leg Message-ID: <24144481.post@talk.nabble.com> Hello, I am using Nibblebill to bill bridged calls initiated via API. The problem is that the billing on the B-leg is only done when the call is terminated and not at heartbeat. For the a-leg it is working. I added the global_heartbeat variable at the b-leg but without success. I use now: originate {ignore_early_media=true,nibble_account=100,nibble_rate=0.02}sofia/external/12345 at serverip &bridge({global_heartbeat=50,nibble_account=100,nibble_rate=0.02}sofia/external/45678 at serverip) thx, MdM -- View this message in context: http://www.nabble.com/Nibblebill-heartbeat-on-B-leg-tp24144481p24144481.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 22 06:22:24 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 09:22:24 -0400 Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile In-Reply-To: <6EA6ABD53A184F1F90CF51F883729E8E@test> References: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> <6EA6ABD53A184F1F90CF51F883729E8E@test> Message-ID: bingo! :P /b On Jun 22, 2009, at 4:06 AM, Darren Schreiber wrote: > Ignore this thread. Apparently I was stripping sip: from the prefix. > I guess you have to specify sip: before utilizing fs_nat and fs_path > variables. > > My bad. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/a0a128e8/attachment-0001.html From brian at freeswitch.org Mon Jun 22 06:23:13 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 09:23:13 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED In-Reply-To: <24143545.post@talk.nabble.com> References: <24143545.post@talk.nabble.com> Message-ID: <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> I'm going to guess you're calling a registered user? If so replace the @ with % /b On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: > > Hi, > > API CALL [originate sofia/external/1001 at 116.50.456.212] > -ERR SERVICE_NOT_IMPLEMENTED > > I receiving this error i dont know y? Can u help mo on this? > > I dialing a mobile number on this sometimes it works... Sometimes it > destroys the call [CALL_DESTROY] > > > Thanks Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/4d946fc7/attachment.html From brian at freeswitch.org Mon Jun 22 06:23:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 09:23:40 -0400 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <8ccbff060906220110y2b9431b5g797d4122152891d9@mail.gmail.com> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> <8ccbff060906220110y2b9431b5g797d4122152891d9@mail.gmail.com> Message-ID: <96FA3310-7670-4540-8F0F-CE1EB2F5D1D1@freeswitch.org> Please open a jira about this. /b On Jun 22, 2009, at 4:10 AM, Dome Charoenyost wrote: > default_language still don't work wirh say > but sound_prefix work fine. > example my javascript > ------------------------------- > session.execute("set", "sound_prefix=/opt/freeswitch/sounds/th"); > session.execute("say","th number pronounced 1346523"); > session.execute("say","th number pronounced 21"); > session.execute("say","th number pronounced 11"); > session.execute("say","th number pronounced 101"); > > > How to check in mod_say_th back to freeswotch ? > > Dome C. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/06c3f15e/attachment.html From JCasale at activenetwerx.com Mon Jun 22 07:36:30 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 14:36:30 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> Message-ID: >It's failing to build the core library. There should be some warning >before it tried to build the modules in the log. You'll have to bear with me, I am not sure exactly what part of the build that is, I see this in the log: +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +----------------------------------------------+ followed by this: +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + ------------------------------------ + and finally it ends with: Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/freeswitch-1.0.4-1-root-rpmbuilder RPM build errors: Which doesn't help:) Thanks! jlc From brian at freeswitch.org Mon Jun 22 07:45:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 10:45:02 -0400 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> Message-ID: <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> Scroll up and look for the real error. /b On Jun 22, 2009, at 10:36 AM, Joseph L. Casale wrote: > You'll have to bear with me, I am not sure exactly what part of the > build > that is, I see this in the log: > > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/bin/make install + > +----------------------------------------------+ > > followed by this: > > +-------- FreeSWITCH install Complete ----------+ > + FreeSWITCH has been successfully installed. + > + + > + Install sounds: + > + (uhd-sounds includes hd-sounds, sounds) + > + (hd-sounds includes sounds) + > + ------------------------------------ + > > and finally it ends with: > > Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/ > freeswitch-1.0.4-1-root-rpmbuilder > > > RPM build errors: > > Which doesn't help:) > Thanks! > jlc Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/7886118d/attachment.html From max.bridgewater at gmail.com Mon Jun 22 08:57:18 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 22 Jun 2009 11:57:18 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Message-ID: Hi Mike, Unfortunately this doesn't seem to solve my problem. Here is my extension again: I've copied it now under: /user/local/freeswitch/conf/dialplan/default /user/local/freeswitch/conf/dialplan/public The different dial strings i tried: "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 & park()" "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67" "{origination_caller_id_number=120003}sofia/internal/242424" "{origination_caller_id_number=120003}sofia/internal/242424%192.168.50.62" My goal: have the call captured by the above extension and redirected to a server socket running at 192.168.50.67:10000. Any thought? Max. On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater wrote: > I don't have my settings to try it right now. Still i have a question. If > it's the way you describe it, why wouldn't sofia/extenal/foo at bar solve the > problem? I think i even copied the extension both to the default directory. > But i will confirm and let you know. > > Max. > > > On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins wrote: > >> Now I feel stupid because I didn't read your original post closely >> enough. >> >> You've defined your "mysocket" extension in the "public" context but when >> you do an origination with sofia/internal/foo at bar it will use the >> "default" context. I think the quickest way to handle this is to create a >> copy of your mysocket.xml file and put it in conf/dialplan/default/ and be >> done with it. >> >> -MC >> >> >> On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Mike, >>> >>> Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML >>> to me though. >>> >>> Max. >>> >>> >>> On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: >>> >>>> Max, >>>> that pastebin failed miserably as none of the xml shows up. can you try >>>> again or use our pastebin.freeswitch.org site? >>>> -MC >>>> >>>> >>>> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >>>> max.bridgewater at gmail.com> wrote: >>>> >>>>> Hi Mike, >>>>> >>>>> It's pasted here: http://pastebin.ca/1466521 >>>>> >>>>> Thanks, >>>>> Max. >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >>>>> >>>>>> Can you turn on debugging (F8) and capture all the output after your >>>>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>>>> -MC >>>>>> >>>>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>>>> max.bridgewater at gmail.com> wrote: >>>>>> >>>>>>> Any help our there? >>>>>>> >>>>>>> I'm still trying to get this piece working. Essentially what i wan to >>>>>>> do is, when a call comes in (from registered devices as well as unregistered >>>>>>> devices), notify the my server socket. Somehow it's not working. The change >>>>>>> i made compared to the standard Freeswitch settings are the following: >>>>>>> >>>>>>> 1) Added following extension that in >>>>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2) Changed file: >>>>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> I noticed that with this extension, all calls received from external >>>>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>>>> But calls from registered devices and initiated using the socket interface >>>>>>> are not forwarded. Is there something that need to be changed in the >>>>>>> profiles? >>>>>>> >>>>>>> or is something wrong with my dial string? >>>>>>> {origination_caller_id_number=12000}sofia/internal/ >>>>>>> 242424 at 192.168.1.62. >>>>>>> >>>>>>> In the logs, i cannot see that that my extension is being matched. >>>>>>> >>>>>>> Any idea, >>>>>>> >>>>>>> Max. >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/4cb9649c/attachment-0001.html From brian at freeswitch.org Mon Jun 22 09:06:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 12:06:25 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Message-ID: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> originate {origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 &socket(192.168.50.67:10000 full) /b On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote: > Hi Mike, > > Unfortunately this doesn't seem to solve my problem. Here is my > extension again: > > > > > > > > I've copied it now under: > > /user/local/freeswitch/conf/dialplan/default > /user/local/freeswitch/conf/dialplan/public > > The different dial strings i tried: > > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 > & park()" > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 > " > "{origination_caller_id_number=120003}sofia/internal/242424" > "{origination_caller_id_number=120003}sofia/internal/ > 242424%192.168.50.62" > > My goal: have the call captured by the above extension and > redirected to a server socket running at 192.168.50.67:10000. > > Any thought? > > Max. > > On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater > wrote: > I don't have my settings to try it right now. Still i have a > question. If it's the way you describe it, why wouldn't sofia/ > extenal/foo at bar solve the problem? I think i even copied the > extension both to the default directory. But i will confirm and let > you know. > > Max. > > > On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins > wrote: > Now I feel stupid because I didn't read your original post closely > enough. > > You've defined your "mysocket" extension in the "public" context but > when you do an origination with sofia/internal/foo at bar it will use > the "default" context. I think the quickest way to handle this is to > create a copy of your mysocket.xml file and put it in conf/dialplan/ > default/ and be done with it. > > -MC > > > On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater > wrote: > Mike, > > Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very > XML to me though. > > Max. > > > On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins > wrote: > Max, > that pastebin failed miserably as none of the xml shows up. can you > try again or use our pastebin.freeswitch.org site? > -MC > > > On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater > wrote: > Hi Mike, > > It's pasted here: http://pastebin.ca/1466521 > > Thanks, > Max. > > > > > On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins > wrote: > Can you turn on debugging (F8) and capture all the output after your > originate? Put it into a pastebin. (pastebin.freeswitch.org) > -MC > > On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater > wrote: > Any help our there? > > I'm still trying to get this piece working. Essentially what i wan > to do is, when a call comes in (from registered devices as well as > unregistered devices), notify the my server socket. Somehow it's not > working. The change i made compared to the standard Freeswitch > settings are the following: > > 1) Added following extension that in /usr/local/freeswitch/conf/ > dialplan/public/mysocket.xml: > > > > > > > > > > > 2) Changed file: /usr/local/freeswitch/conf/autoload_configs/ > event_socket.conf to: > > > > > > > > > > > > > I noticed that with this extension, all calls received from external > providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my > socket. But calls from registered devices and initiated using the > socket interface are not forwarded. Is there something that need to > be changed in the profiles? > > or is something wrong with my dial string? {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62 > . > > In the logs, i cannot see that that my extension is being matched. > > Any idea, > > Max. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b9ef667a/attachment.html From max.bridgewater at gmail.com Mon Jun 22 10:08:51 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 22 Jun 2009 13:08:51 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> Message-ID: Hmm thamks. I tried it and it doesn't work out of the box. Here are my logs: http://pastebin.freeswitch.org/9454 Thanks, Max. On Mon, Jun 22, 2009 at 12:06 PM, Brian West wrote: > originate > {origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67<%7Borigination_caller_id_number=120003%7Dsofia/internal/242424 at 192.168.50.67>&socket( > 192.168.50.67:10000 full) > /b > > On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote: > > Hi Mike, > > Unfortunately this doesn't seem to solve my problem. Here is my extension > again: > > > > /> > > > > I've copied it now under: > > /user/local/freeswitch/conf/dialplan/default > /user/local/freeswitch/conf/dialplan/public > > The different dial strings i tried: > > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67& park()" > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67" > "{origination_caller_id_number=120003}sofia/internal/242424" > "{origination_caller_id_number=120003}sofia/internal/242424%192.168.50.62" > > My goal: have the call captured by the above extension and redirected to a > server socket running at 192.168.50.67:10000. > > Any thought? > > Max. > > On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> I don't have my settings to try it right now. Still i have a question. If >> it's the way you describe it, why wouldn't sofia/extenal/foo at bar solve >> the problem? I think i even copied the extension both to the default >> directory. But i will confirm and let you know. >> >> Max. >> >> >> On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins wrote: >> >>> Now I feel stupid because I didn't read your original post closely >>> enough. >>> >>> You've defined your "mysocket" extension in the "public" context but when >>> you do an origination with sofia/internal/foo at bar it will use the >>> "default" context. I think the quickest way to handle this is to create a >>> copy of your mysocket.xml file and put it in conf/dialplan/default/ and be >>> done with it. >>> >>> -MC >>> >>> >>> On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater < >>> max.bridgewater at gmail.com> wrote: >>> >>>> Mike, >>>> >>>> Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML >>>> to me though. >>>> >>>> Max. >>>> >>>> >>>> On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: >>>> >>>>> Max, >>>>> that pastebin failed miserably as none of the xml shows up. can you try >>>>> again or use our pastebin.freeswitch.org site? >>>>> -MC >>>>> >>>>> >>>>> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >>>>> max.bridgewater at gmail.com> wrote: >>>>> >>>>>> Hi Mike, >>>>>> >>>>>> It's pasted here: http://pastebin.ca/1466521 >>>>>> >>>>>> Thanks, >>>>>> Max. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins >>>>> > wrote: >>>>>> >>>>>>> Can you turn on debugging (F8) and capture all the output after your >>>>>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>>>>> -MC >>>>>>> >>>>>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>>>>> max.bridgewater at gmail.com> wrote: >>>>>>> >>>>>>>> Any help our there? >>>>>>>> >>>>>>>> I'm still trying to get this piece working. Essentially what i wan >>>>>>>> to do is, when a call comes in (from registered devices as well as >>>>>>>> unregistered devices), notify the my server socket. Somehow it's not >>>>>>>> working. The change i made compared to the standard Freeswitch settings are >>>>>>>> the following: >>>>>>>> >>>>>>>> 1) Added following extension that in >>>>>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> > >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> 2) Changed file: >>>>>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> I noticed that with this extension, all calls received from external >>>>>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>>>>> But calls from registered devices and initiated using the socket interface >>>>>>>> are not forwarded. Is there something that need to be changed in the >>>>>>>> profiles? >>>>>>>> >>>>>>>> or is something wrong with my dial string? >>>>>>>> {origination_caller_id_number=12000}sofia/internal/ >>>>>>>> 242424 at 192.168.1.62. >>>>>>>> >>>>>>>> In the logs, i cannot see that that my extension is being matched. >>>>>>>> >>>>>>>> Any idea, >>>>>>>> >>>>>>>> Max. >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/6f165662/attachment-0001.html From brian at freeswitch.org Mon Jun 22 10:18:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 13:18:44 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> Message-ID: <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> what is 242424? If its a locally registered user you should be using a % instead of an @ /b On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: > Hmm thamks. I tried it and it doesn't work out of the box. Here are > my logs: http://pastebin.freeswitch.org/9454 > > Thanks, > Max. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b9ebc105/attachment.html From lon at kickasspixels.com Mon Jun 22 10:18:40 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 22 Jun 2009 10:18:40 -0700 Subject: [Freeswitch-users] Limit length of call with mod_limit? Message-ID: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> Hi there, Can mod_limit be used to restrict the length of a single call? I checked the wiki, dug into the code of mod_limit this weekend and couldn't find an answer. Lon From brian at freeswitch.org Mon Jun 22 10:22:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 13:22:11 -0400 Subject: [Freeswitch-users] Limit length of call with mod_limit? In-Reply-To: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> References: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Channel_Variables /b On Jun 22, 2009, at 1:18 PM, Lon Baker wrote: > Hi there, > > Can mod_limit be used to restrict the length of a single call? > > I checked the wiki, dug into the code of mod_limit this weekend and > couldn't find an answer. > > Lon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/deeed857/attachment.html From raul at etellicom.com Mon Jun 22 10:44:18 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 22 Jun 2009 14:44:18 -0300 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: Message-ID: <1245692658.5598.19.camel@raul-laptop> I have recently updated the RPM spec for FreeSWITCH to use the latest SVN release, including some unspecified files for the newest mods (with nibblebil, unimrcp, etc, just like your build output is bitching about). Follow these steps to build it: 1 - Get the latest SVN release and make a tar-ball for it: $ cd /usr/src/redhat/SOURCES/ $ svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4trunk $ tar -cjvf freeswitch-1.0.4trunk.tar.bz2 freeswitch-1.0.4trunk/ 2 - Grab the libraries required to build FS: $ wget -q -O - http://www.etellicom.com/~raul/freeswitch_deps.txt | bash 3 - Download the RPM spec: $ cd /usr/src/redhat/SPECS/ $ wget http://www.etellicom.com/~raul/freeswitch.spec 4 - Build it :) $ rpmbuild -ba freeswitch.spec It works fine with a standard CentOS 5.3 installation. Regards, Raul On Mon, 2009-06-22 at 01:25 +0000, Joseph L. Casale wrote: > I attempted to build rpm's from the included spec file using a non-root user > build environment. The steps I used are as follows: > > 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS > 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as expected by the spec: > svn co http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4 > tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ > 3. Copy spec to SPECS directory: > 4. Pull in Source (Source0 doesn't exist yet, I make it above): > for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done > 5. Check spec in svn copy for deps: > yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' freeswitch.spec) > 6.Build rpm: > rpmbuild -ba freeswitch.spec > > After some time, near the end I see various issues like the following: > > making install mod_speex > installing mod_speex.so > quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/freeswitch-1.0.4/libfreeswitch.la' > has not been installed in `/opt/freeswitch/lib' > > > It also seems to download everything it would normally again, then fails with > several errors like the following: > > RPM build errors: > File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/opt/freeswitch/mod/ozmod_wanpipe.so* > Installed (but unpackaged) file(s) found: > /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml > /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml > /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml > /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml > . > . > . > > After which no rpm's are built? Anyone know what tricks are still needed with > the spec from svn? > > Thanks! > jlc > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From max.bridgewater at gmail.com Mon Jun 22 11:01:29 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 22 Jun 2009 14:01:29 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> References: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> Message-ID: It's nothing. There is no extension like that. Shouldn't this nonetheless be caught by a regex such as the following? field="destination_number" expression="^242.*" The issue i have here is that it seems that the extensions aren't even processed. Usually, the log would show the list of processed extensions, each prefixed with the result "PASS", "FAIL". Max. On Mon, Jun 22, 2009 at 1:18 PM, Brian West wrote: > what is 242424? If its a locally registered user you should be using a % > instead of an @ > /b > > On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: > > Hmm thamks. I tried it and it doesn't work out of the box. Here are my > logs: http://pastebin.freeswitch.org/9454 > > Thanks, > Max. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/1345f913/attachment.html From JCasale at activenetwerx.com Mon Jun 22 11:50:56 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 18:50:56 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> Message-ID: >Scroll up and look for the real error. All I see are these: *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libedit/src/.libs/libedit.a is not portable! ****** Integer sample type enabled ****** *** The gtk-config script installed by GTK could not be found *** If GTK was installed in PREFIX, make sure PREFIX/bin is in *** your path, or set the GTK_CONFIG environment variable to the *** full path to gtk-config. I am pulling down the newer libraries and updated spec now to try that. Thanks guys! jlc From brian at freeswitch.org Mon Jun 22 11:57:45 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 14:57:45 -0400 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> Message-ID: <3CE93178-B052-4CCC-BE33-3DBA79661956@freeswitch.org> Ok, thats not the issue... look lower or post the full log. /b On Jun 22, 2009, at 2:50 PM, Joseph L. Casale wrote: > > All I see are these: > > *** Warning: Linking the shared library libfreeswitch.la against the > *** static library libs/libedit/src/.libs/libedit.a is not portable! > ****** Integer sample type enabled ****** > > *** The gtk-config script installed by GTK could not be found > *** If GTK was installed in PREFIX, make sure PREFIX/bin is in > *** your path, or set the GTK_CONFIG environment variable to the > *** full path to gtk-config. > > I am pulling down the newer libraries and updated spec now to try > that. > > Thanks guys! > jlc Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/26bca5c1/attachment-0001.html From timb0311 at hotmail.com Mon Jun 22 12:01:03 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 22 Jun 2009 15:01:03 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: Just wanted to follow-up again. Is this the proper or best way to configure this? See below... Tim From: timb0311 at hotmail.com To: freeswitch-users at lists.freeswitch.org Subject: (Found Fix) Transmit fax locally for test Date: Fri, 19 Jun 2009 23:06:10 -0400 Ok so after many attempts of trial and error I narrowed it down to acls. So when trying to orginate a call to the local FS extension it was getting blocked. Adding the following allow with my freeswitch IP to the domains list allowed the originate to take place. acl.conf.xml: So now this statement works for local fax testing: originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) Now my question is, is this the proper or best way to configure this? Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503 at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000 at x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users at lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > > haha, yeah i see it now... duh. pulled an all nighter, too many things > going on. must have overlooked it. > > > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > > the login and password is pastebin/freeswitch > > > > been trying to break myself into freeswitch on top of my original workload. > thanks for the help. > Bing? brings you maps, menus, and reviews organized in one place. Try it now. _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing? now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try bing_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/40b5079d/attachment.html From brian at freeswitch.org Mon Jun 22 12:05:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 15:05:20 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67@freeswitch.org> what is 8000? is it local or is it a remote endpoint? /b On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > test.tif) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/f773841e/attachment.html From drago at windstream.net Mon Jun 22 13:33:51 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 22 Jun 2009 16:33:51 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: <004e01c9f378$c1707e90$44517bb0$@net> Michael, It is definitely a problem in this build (13754M) (Windows) I found build from March and it does not brakes (still no joy, but not sure for now if the exchange side is configured properly). Again, I would submit bug report is you can point me to instructions how to do collect backtrace on Windows OS. Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Sunday, June 21, 2009 4:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: After days of running around clueless, I have no other option but to ask the community for help one last time? Here is what happens: 1. FS sends INVITE 2. Exchange answers with ?302 Moved Temporarily? 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: ?MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The ?From? field?s address to be in the format ??; otherwise, MS Exchange drops the call.? I don?t know if this is the only problem? However, I see exactly this behavior: ?PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established.? After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/23f8e33f/attachment-0001.html From msc at freeswitch.org Mon Jun 22 13:44:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 22 Jun 2009 15:44:20 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> Message-ID: <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> On Mon, Jun 22, 2009 at 1:01 PM, Max Bridgewater wrote: > It's nothing. There is no extension like that. Shouldn't this nonetheless > be caught by a regex such as the following? > > field="destination_number" expression="^242.*" > > The issue i have here is that it seems that the extensions aren't even > processed. Usually, the log would show the list of processed extensions, > each prefixed with the result "PASS", "FAIL". > Max, if your originate line already has the sofia dialstring then there's really no reason to send the call through the dialplan - it already knows where to go. If you want to force the call through the dialplan then use loopback. However, you need some sort of endpoint for that to work. In your example you have this originate line: originate {origination_caller_id_number=120003}sofia/internal/ 242424 at 192.168.50.67 &socket(192.168.50.67:10000 full) Is 242424 at 192.168.50.67 a locally registered user? If so you could just do this: originate {origination_caller_id_number=120003} loopback/242424 &socket( 192.168.50.67:10000 full) This would run the A leg through the dialplan to look for destination number "242424" and then handle appropriately. If I understand your scenario I believe you are trying to get one leg of the call established and then the other leg handled by the event socket. What is the endpoint you want handled? A SIP phone that is registered locally? Or something else? In any case, you can CAN loop it through the dialplan but you aren't forced to do so. Assuming 1000 is locally registered: originate {origination_caller_id_number=120003} sofia/internal/1000%192.168.50.67 &socket(192.168.50.67:10000 full) originate {origination_caller_id_number=120003} user/1000 &socket( 192.168.50.67:10000 full) originate {origination_caller_id_number=120003} loopback/1000 &socket( 192.168.50.67:10000 full) NOTE: the first two do not use the dialplan but the third example does. This means you MUST handle destination_number="1000" in your dialplan (which the default config does). Hope this helps. -MC > Max. > > On Mon, Jun 22, 2009 at 1:18 PM, Brian West wrote: > >> what is 242424? If its a locally registered user you should be using a % >> instead of an @ >> /b >> >> On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: >> >> Hmm thamks. I tried it and it doesn't work out of the box. Here are my >> logs: http://pastebin.freeswitch.org/9454 >> >> Thanks, >> Max. >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/2f970056/attachment.html From larclap at yahoo.com Mon Jun 22 13:49:16 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 22 Jun 2009 13:49:16 -0700 Subject: [Freeswitch-users] Polycom configuration problems? Message-ID: <003f01c9f37a$e8b70d00$ba252700$@com> I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/a7c44c1e/attachment-0001.html From chris at cloudtel.com Mon Jun 22 14:57:23 2009 From: chris at cloudtel.com (Chris Burns) Date: Mon, 22 Jun 2009 16:57:23 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <003f01c9f37a$e8b70d00$ba252700$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> Message-ID: Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or .digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 > lines on the phone. The first two are registered with a SwitchVox, the last > with Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/cb52cac0/attachment.html From timb0311 at hotmail.com Mon Jun 22 15:37:47 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 22 Jun 2009 18:37:47 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: 8000 is a local extension defined in the default dialplan. Tim > ------------------------------ > > Message: 2 > Date: Mon, 22 Jun 2009 15:05:20 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > what is 8000? is it local or is it a remote endpoint? > > /b > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > test.tif) > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > _________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/5ab527bb/attachment.html From jim at evolutiontel.net Mon Jun 22 15:53:22 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 23 Jun 2009 08:53:22 +1000 Subject: [Freeswitch-users] Limit length of call with mod_limit? In-Reply-To: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> References: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> Message-ID: This line in your diaplan will set a timer to hangup the calls x secs after answer. On Tue, Jun 23, 2009 at 3:18 AM, Lon Baker wrote: > Hi there, > > Can mod_limit be used to restrict the length of a single call? > > I checked the wiki, dug into the code of mod_limit this weekend and > couldn't find an answer. > > Lon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/d0c15ac1/attachment.html From JCasale at activenetwerx.com Mon Jun 22 15:55:17 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 22:55:17 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: <1245692658.5598.19.camel@raul-laptop> References: <1245692658.5598.19.camel@raul-laptop> Message-ID: >I have recently updated the RPM spec for FreeSWITCH to use the latest >SVN release, including some unspecified files for the newest mods (with >nibblebil, unimrcp, etc, just like your build output is bitching about). >Follow these steps to build it: Raul, Appreciate this, it worked. I am re-running it as the first time I was only logging stdout and it appears there are some errors in my build worth exploring. Nothing prevented the build, so I created a local repo with all the rpm's and executed a `yum install freeswitch` and I see that it never pulled in anything else. Where abouts in the docs could I find info on knowing what I need for an initial install to test? Thanks for the help! jlc From jingwei.yang at gmail.com Mon Jun 22 19:33:09 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 23 Jun 2009 10:33:09 +0800 Subject: [Freeswitch-users] How to originate gtalk calls Message-ID: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch> *originate dingaling/gmail.com/userAAA at gmail.com &echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* Please kindly let me know what the correct originate string is. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/a8b8d86b/attachment.html From brian at freeswitch.org Mon Jun 22 20:10:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 22:10:42 -0500 Subject: [Freeswitch-users] How to originate gtalk calls In-Reply-To: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Message-ID: <5AB2BF21-119D-478E-B582-229E1AD91D1B@freeswitch.org> Might need to compile and load mod_dingaling first. /b On Jun 22, 2009, at 9:33 PM, Jingwei Yang wrote: > Hi Guys, > > I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ > . > > But i'm not sure how to originate calls to different gtalk users > dynamically. I've tried this: > > freeswitch> originate dingaling/gmail.com/userAAA at gmail.com &echo > > but got CHAN_NOT_IMPLEMENTED error. > > 2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot > create outgoing channel of type [dingaling] cause: > [CHAN_NOT_IMPLEMENTED] > > Please kindly let me know what the correct originate string is. > Thanks! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/4a47413e/attachment.html From mashudiflexi at telkom.co.id Mon Jun 22 20:28:07 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Tue, 23 Jun 2009 10:28:07 +0700 Subject: [Freeswitch-users] video playback on FS In-Reply-To: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Message-ID: <4A404BC7.2060403@telkom.co.id> Dear All, on the default.xml in dialplan directory of FS, content video extension dialplan with file extension fsv, 562 563 564 565 566 567 568 569 570 571 572 573 574 what is the fsv video format from? as we know flv for flash video, how to convert from mp4 or avi to fsv file extension? thank you in advanced, best regard, mashudi ==================================== Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya: - hubungi 147 - http://www.telkomflexi.com - ketik INFO, sms ke 345. From jmesquita at gmail.com Mon Jun 22 20:14:43 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 23 Jun 2009 00:14:43 -0300 Subject: [Freeswitch-users] How to originate gtalk calls In-Reply-To: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Message-ID: <5a8712120906222014s43c220b0x29ff0c3e6e790fb0@mail.gmail.com> try load mod_dingaling. If that does not work, get to the source dir, edit modules.conf, uncomment mod_dingaling, make && make install Dont forget to load the mod once FS is up again.. jmesquita On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang wrote: > Hi Guys, > > I've configured a gtalk client based on the steps in this url: > http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. > > But i'm not sure how to originate calls to different gtalk users > dynamically. I've tried this: > > freeswitch> *originate dingaling/gmail.com/userAAA at gmail.com &echo* > > but got CHAN_NOT_IMPLEMENTED error. > > *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create > outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* > > Please kindly let me know what the correct originate string is. Thanks! > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/3fc8b1fc/attachment.html From jingwei.yang at gmail.com Mon Jun 22 20:25:08 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 23 Jun 2009 11:25:08 +0800 Subject: [Freeswitch-users] How to originate gtalk calls In-Reply-To: <5a8712120906222014s43c220b0x29ff0c3e6e790fb0@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> <5a8712120906222014s43c220b0x29ff0c3e6e790fb0@mail.gmail.com> Message-ID: <13529f9d0906222025v781dbc80qd3ca9d1179730282@mail.gmail.com> Hi Brian and Jo?o, you're right, I forgot to load mod_dingaling. Thanks for the help. 2009/6/23 Jo?o Mesquita > try load mod_dingaling. > > If that does not work, get to the source dir, edit modules.conf, uncomment > mod_dingaling, make && make install > > Dont forget to load the mod once FS is up again.. > > jmesquita > > On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang wrote: > >> Hi Guys, >> >> I've configured a gtalk client based on the steps in this url: >> http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. >> >> But i'm not sure how to originate calls to different gtalk users >> dynamically. I've tried this: >> >> freeswitch> *originate dingaling/gmail.com/userAAA at gmail.com &echo* >> >> but got CHAN_NOT_IMPLEMENTED error. >> >> *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot >> create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED] >> * >> >> Please kindly let me know what the correct originate string is. Thanks! >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/7e47b81b/attachment.html From brian at freeswitch.org Mon Jun 22 20:26:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 22:26:18 -0500 Subject: [Freeswitch-users] video playback on FS In-Reply-To: <4A404BC7.2060403@telkom.co.id> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> <4A404BC7.2060403@telkom.co.id> Message-ID: <7B3175DB-042E-45BB-BD87-7DC5573B8622@freeswitch.org> On Jun 22, 2009, at 10:28 PM, mashudi wrote: > Dear All, > on the default.xml in dialplan directory of FS, content video > extension > dialplan with file extension fsv, > > 562 > 563 expression="^9993$"> > 564 > 565 data="/tmp/testrecord.fsv"/> > 566 > 567 > 568 > 569 > 570 expression="^9994$"> > 571 > 572 > 573 > 574 > > what is the fsv video format from? as we know flv for flash video, > how to convert from mp4 or avi to fsv file extension? We just save the raw RTP and stream it back out... btw don't hijack threads please. > thank you in advanced, > > best regard, > > mashudi Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/36b08c36/attachment-0001.html From darklion11 at yahoo.com Mon Jun 22 20:38:24 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 20:38:24 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED In-Reply-To: <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> References: <24143545.post@talk.nabble.com> <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> Message-ID: <24158819.post@talk.nabble.com> Nope. I just want to call a mobile number with no register number. Brian West-3 wrote: > > I'm going to guess you're calling a registered user? If so replace > the @ with % > > /b > > On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: > >> >> Hi, >> >> API CALL [originate sofia/external/1001 at 116.50.456.212] >> -ERR SERVICE_NOT_IMPLEMENTED >> >> I receiving this error i dont know y? Can u help mo on this? >> >> I dialing a mobile number on this sometimes it works... Sometimes it >> destroys the call [CALL_DESTROY] >> >> >> Thanks > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24158819.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 22 21:02:49 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 21:02:49 -0700 (PDT) Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 Message-ID: <24158823.post@talk.nabble.com> When I calling an outbound extension it appears: name is FreeSWITCH and number is 000000000 How can i change it depends on the user who is calling? Sample 1001->64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From q.edward at gmail.com Mon Jun 22 21:26:32 2009 From: q.edward at gmail.com (Edward Q.) Date: Tue, 23 Jun 2009 00:26:32 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <24158823.post@talk.nabble.com> References: <24158823.post@talk.nabble.com> Message-ID: <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: > > When I calling an outbound extension it appears: > > name is FreeSWITCH and number is 000000000 > > How can i change it depends on the user who is calling? > > Sample 1001->64521223 > > I just want the name 1001 to appear not FreeSWITCH same as the number > > Thanks > > > > -- > View this message in context: > http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/176d1862/attachment.html From otrcomm at isp-systems.net Mon Jun 22 20:31:54 2009 From: otrcomm at isp-systems.net (murrah boswell) Date: Mon, 22 Jun 2009 20:31:54 -0700 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment Message-ID: <4A404CAA.5080809@isp-systems.net> Hello All, I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance on how to setup a testbed in a thin client environment. I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and utilize fl_teachertool 0.07 to monitor the connected terminal clients (TCs). If you are not familiar with fl_teachertool, it allows a teacher to view thumbnail images of each TC logged in to the server. The teacher can click on any thumbnail and enlarge the view, monitor all applications running on a given TC, and take control of the keyboard and mouse of the TC. These are just a few of the capabilities of fl_teachertool. What I want to do is allow the teacher to establish voice communication using headsets and microphones with any one of the TCs by making a "phone" call via ethernet based upon ip of the TC through freeswitch using a softphone. Does this sound like something that is possible using freeswitch? If so, could someone please give me very basic instructions on how to setup this proof of concept? If I can just get a "teacher" stationed at my server talking to one "student" at a TC, I believe I can go from there. Currently I have a voiper softphone that functions, I believe, under gnome, but I have no idea how to configure the voiper to initiate calls through freeswitch or how to configure freeswitch to route the call to one of my TCs. I also need to keep this system fully self contained. That is, I can not have a requirement to use an outside sip service provider. Also, I would use any other linux sip softphones known to work with freeswitch that people feel would work better than a voiper. voiper seems to be more windows and mac based. I would really like to use an ekiga since they seem to be more linux based, but I do not believe that they have been thoroughly tested with freeswitch. Any help would be greatly appreciated! Regards, Murrah Boswell From vince.freeswitch at hightek.org Mon Jun 22 21:08:53 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Mon, 22 Jun 2009 23:08:53 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD Message-ID: <20090623040853.GA84157@quark.hightek.org> Hi. I have been searching for an alternative PBX to asterisk (which has not been all that stable) to run on Dragonfly BSD. I spent a fair amount of time a couple months ago trying to compile freeswitch without success. I have since tried Yate, but it consumes 85-95% of the cpu when idle (not processing any calls). I am considering revisiting freeswitch. I have included below my notes of all the various problems I encountered before, which ones I resolved and how, up to the point where I left off. I am in hopes of getting feedback on whether any of the issues have been fixed or are planned to be fixed and/or suggestions on getting it working. Also, perhaps these notes on my experiences will be helpful for the developers to improve freeswitch. Keep in mind that Dragonfly is a branch from Freebsd and stays fairly compatible. I am able to compile most software, that is ported to FreeBSD, with few problems. Here are my notes ================= To compile on dragonfly BSD 1.10.1-RELEASE ========================================== I had to add -D__FreeBSD__ to CPPFLAGS ln sh to bash or zsh because I got "unexpected operator" errors from "test" during configure with the bsd shell. ln make to gmake Their scripts were calling make even though I ran the build using gmake. Must have apr-0.9.16.2.0.61 and apr-util-0.9.16.2.0.61 installed apr-0.xxx has headers, which freeswitch is including, that apr-1.xxx does not have. I sym-linked the apr-util libs from apr-util-1.2.8nb1 because apr-util-0.9.16.2.0.61 was not available as a binary package. The freeswitch-1.0.3.tar.gz release did not have bootstrap.sh, which is required for building from the svn repository. Somebody on the #freeswitch IRC suggested I get it from the subversion repository and run the bootstrap script. bootstrap.sh is not in the release. ========================================== Tue Mar 31 00:12:37 CDT 2009 I posted on the freeswitch mailing list asking about the compilation errors Got no responses. This first set of apr_... warnings turned out to clearly be from not having the correct apr-util package installed, I should have gotten a response on the list about it, considering it is a clear dependency that they do not directly specify on the web site or the source docs. apr-util is a dependency of subversion. They list SVN as a dependency of freeswitch, which is the utility in the subversion package, not the package name. svn should not be a dependency to build from a release archive that is not retrieved from the svn repository. As it turns out, it had to be apr-util version 0.9.15. See notes below. ========================================== Tue Mar 31 22:22:58 CDT 2009 I tried the freeswitch-snapshot.tar.gz from the freeswitch site, which is a 03/30/2009 snapshot from the svn trunk. *concern* It was nearly twice as big as the release for some reason. 27016871 Mar 28 13:08 freeswitch-1.0.3.tar.gz 52854882 Mar 31 18:25 freeswitch-snapshot.tar.gz Running bootstrap.sh produced a bunch of these errors from automake: Use of uninitialized value in exists at /usr/pkg/bin/automake line 4823, line 1. Use of uninitialized value in concatenation (.) or string at /usr/pkg/bin/automake line 4823, line 1. automake: #################### automake: ## Internal Error ## automake: #################### automake: unrequested trace `' automake: Please contact . at /usr/pkg/share/automake-1.10/Automake/Channels.pm line 570 Automake::Channels::msg('automake', '', 'unrequested trace `\'') called at /usr/pkg/share/automake-1.10/Automake/ChannelDefs.pm line 191 Automake::ChannelDefs::prog_error('unrequested trace `\'') called at /usr/pkg/bin/automake line 4823 Automake::scan_autoconf_traces('configure.ac') called at /usr/pkg/bin/automake line 5046 Automake::scan_autoconf_files() called at /usr/pkg/bin/automake line 781 ========================================== Thu Apr 2 20:51:44 CDT 2009 Going back to the 1.0.3 release. I was getting a a bunch of apr_... warnings like Compiling src/switch_apr.c ... src/switch_apr.c: In function `switch_thread_self': src/switch_apr.c:74: warning: implicit declaration of function `apr_os_thread_current' src/switch_apr.c:74: warning: return makes pointer from integer without a cast ... Then gmake[3]: *** [libfreeswitch_la-switch_apr.lo] Error 1 I finally got past that by installing apr-0.9.16.2.0.61 and apr-util-0.9.15. I compiled apr-util-0.9.15 myself because it was not available as a binary package. Source files are including headers from apr-util-0.9.15 that do not exist in the binary package, apr-util-1.2.8nb1.tgz. I also got past the "Cannot guess build type" error during configure by adding "--build=i386" instead of having to use the uname wrapper to fake FreeBSD. ========================================== New error: (This one is not the fault of freeswitch) Compiling src/switch_core.c ... src/switch_core.c: In function `switch_core_setrlimits': src/switch_core.c:795: error: `RLIMIT_AS' undeclared (first use in this function) Turns out RLIMIT_AS is defined in /usr/include/sys/resource.h on some other systems including FreeBSD but not Dragonfly. It is used in the source like this #if !defined(__OpenBSD__) && !defined(__NetBSD__) setrlimit(RLIMIT_AS, &rlp); It is defined in resource.h in FreeBSD like this #define RLIMIT_AS RLIMIT_VMEM /* standard name for RLIMIT_VMEM */ I created a patch for src/switch_core.c which fixed this error. Since then, I discussed this on the Dragonfly BSD mailing list and it is apparently going to be fixed on a future release of Dragonfly now that they are aware of it. ========================================== New error: Tons of warnings like warning: return makes pointer from integer without a cast (this needs cleaned up) Then error'd with gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. I cd'd to libs/sofia-sip/libsofia-sip-ua/su and ran make there. It completed successfully in spite of the same warnings. Then ran make again form the top. Got further until ========================================== New error: making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ports/freeswitch-1.0.3/work/freeswitch-1.0.3/libs/js/nsprpub/dist/include/nspr/.: File exists Why would it be trying to make a sym-link of '.' ?? ========================================== Giving up for now. From harmeet at litatel.com Mon Jun 22 21:46:22 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Tue, 23 Jun 2009 00:46:22 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Message-ID: In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > Edmar > > Eso esta en freeswitch/conf/vars.xml en ese archivo. > If i am not mistaken and anyone welcome to correct me i just told Edmar > this is set in freeswitch/conf/vars.xml ... file > Ed > > > On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: > >> >> When I calling an outbound extension it appears: >> >> name is FreeSWITCH and number is 000000000 >> >> How can i change it depends on the user who is calling? >> >> Sample 1001->64521223 >> >> I just want the name 1001 to appear not FreeSWITCH same as the number >> >> Thanks >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/bf321631/attachment-0001.html From harmeet at litatel.com Mon Jun 22 21:47:46 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Tue, 23 Jun 2009 00:47:46 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Message-ID: Oops, that was too quick! In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - Thanks On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh wrote: > In my case the 1001 resides in - > /usr/local/freeswitch/conf/directory/default/1001.xml > > And you set the Caller Name and ID by adding - > > > > > > > On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > >> Edmar >> >> Eso esta en freeswitch/conf/vars.xml en ese archivo. >> If i am not mistaken and anyone welcome to correct me i just told Edmar >> this is set in freeswitch/conf/vars.xml ... file >> Ed >> >> >> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: >> >>> >>> When I calling an outbound extension it appears: >>> >>> name is FreeSWITCH and number is 000000000 >>> >>> How can i change it depends on the user who is calling? >>> >>> Sample 1001->64521223 >>> >>> I just want the name 1001 to appear not FreeSWITCH same as the number >>> >>> Thanks >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/8a5f05f7/attachment-0001.html From q.edward at gmail.com Mon Jun 22 23:06:42 2009 From: q.edward at gmail.com (Edward Q.) Date: Tue, 23 Jun 2009 02:06:42 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Message-ID: <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> Sorry Edmar I missundertood you .. I thought you wanted to change the number showing once you were going out not the 1001.xml file. In this case Harmeet is right. There you have those values to to make the changes. My bad. Ed On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh wrote: > In my case the 1001 resides in - > /usr/local/freeswitch/conf/directory/default/1001.xml > > And you set the Caller Name and ID by adding - > > > > > > > On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > >> Edmar >> >> Eso esta en freeswitch/conf/vars.xml en ese archivo. >> If i am not mistaken and anyone welcome to correct me i just told Edmar >> this is set in freeswitch/conf/vars.xml ... file >> Ed >> >> >> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: >> >>> >>> When I calling an outbound extension it appears: >>> >>> name is FreeSWITCH and number is 000000000 >>> >>> How can i change it depends on the user who is calling? >>> >>> Sample 1001->64521223 >>> >>> I just want the name 1001 to appear not FreeSWITCH same as the number >>> >>> Thanks >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/b0b39470/attachment.html From darklion11 at yahoo.com Mon Jun 22 23:39:39 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 23:39:39 -0700 (PDT) Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> Message-ID: <24160712.post@talk.nabble.com> Actually the extension_caller_id=Extension 1001 and extension_caller_number=1001 is set as Harmeet says but the same issue FreeSwitch the caller name and the number is 0000000 i just want 1001 the caller number and the id Edmar Edward Q. wrote: > > Sorry Edmar > > I missundertood you .. I thought you wanted to change the number showing > once you were going out not the 1001.xml file. > In this case Harmeet is right. There you have those values to to make the > changes. > My bad. > Ed > > On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh > wrote: > >> In my case the 1001 resides in - >> /usr/local/freeswitch/conf/directory/default/1001.xml >> >> And you set the Caller Name and ID by adding - >> >> >> >> >> >> >> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: >> >>> Edmar >>> >>> Eso esta en freeswitch/conf/vars.xml en ese archivo. >>> If i am not mistaken and anyone welcome to correct me i just told Edmar >>> this is set in freeswitch/conf/vars.xml ... file >>> Ed >>> >>> >>> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz >>> wrote: >>> >>>> >>>> When I calling an outbound extension it appears: >>>> >>>> name is FreeSWITCH and number is 000000000 >>>> >>>> How can i change it depends on the user who is calling? >>>> >>>> Sample 1001->64521223 >>>> >>>> I just want the name 1001 to appear not FreeSWITCH same as the number >>>> >>>> Thanks >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24160712.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Tue Jun 23 04:03:54 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 23 Jun 2009 19:03:54 +0800 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <24160712.post@talk.nabble.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> <24160712.post@talk.nabble.com> Message-ID: <1AEB5722-06B2-4B3F-858B-164E10008C5B@gmail.com> depending how do you make out going call. On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote: > > Actually the extension_caller_id=Extension 1001 and > extension_caller_number=1001 is set as Harmeet says but the same issue > FreeSwitch the caller name and the number is 0000000 i just want > 1001 the > caller number and the id > > Edmar > > Edward Q. wrote: >> >> Sorry Edmar >> >> I missundertood you .. I thought you wanted to change the number >> showing >> once you were going out not the 1001.xml file. >> In this case Harmeet is right. There you have those values to to >> make the >> changes. >> My bad. >> Ed >> >> On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh >> wrote: >> >>> In my case the 1001 resides in - >>> /usr/local/freeswitch/conf/directory/default/1001.xml >>> >>> And you set the Caller Name and ID by adding - >>> >>> >>> >>> >>> >>> >>> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. >>> wrote: >>> >>>> Edmar >>>> >>>> Eso esta en freeswitch/conf/vars.xml en ese archivo. >>>> If i am not mistaken and anyone welcome to correct me i just told >>>> Edmar >>>> this is set in freeswitch/conf/vars.xml ... file >>>> Ed >>>> >>>> >>>> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> When I calling an outbound extension it appears: >>>>> >>>>> name is FreeSWITCH and number is 000000000 >>>>> >>>>> How can i change it depends on the user who is calling? >>>>> >>>>> Sample 1001->64521223 >>>>> >>>>> I just want the name 1001 to appear not FreeSWITCH same as the >>>>> number >>>>> >>>>> Thanks >>>>> >>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24160712.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Tue Jun 23 05:51:30 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 23 Jun 2009 22:51:30 +1000 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? Message-ID: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks! From larclap at yahoo.com Tue Jun 23 06:26:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 23 Jun 2009 06:26:38 -0700 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> Message-ID: <005401c9f406$3ced3710$b6c7a530$@com> I'm sorry Chris, but I don't know where the look for the "global sip.cfg and mac/phone specific cfg" settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or .digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/da144c31/attachment-0001.html From harmeet at litatel.com Tue Jun 23 06:27:34 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Tue, 23 Jun 2009 09:27:34 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <24160712.post@talk.nabble.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> <24160712.post@talk.nabble.com> Message-ID: Check your dialplan where you call "bridge" to gateway to make outgoing calls. Stick in the following lines before the bridge call - On Tue, Jun 23, 2009 at 2:39 AM, Edmar Cruz wrote: > > Actually the extension_caller_id=Extension 1001 and > extension_caller_number=1001 is set as Harmeet says but the same issue > FreeSwitch the caller name and the number is 0000000 i just want 1001 the > caller number and the id > > Edmar > > Edward Q. wrote: > > > > Sorry Edmar > > > > I missundertood you .. I thought you wanted to change the number showing > > once you were going out not the 1001.xml file. > > In this case Harmeet is right. There you have those values to to make the > > changes. > > My bad. > > Ed > > > > On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh > > wrote: > > > >> In my case the 1001 resides in - > >> /usr/local/freeswitch/conf/directory/default/1001.xml > >> > >> And you set the Caller Name and ID by adding - > >> > >> > >> > >> > >> > >> > >> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > >> > >>> Edmar > >>> > >>> Eso esta en freeswitch/conf/vars.xml en ese archivo. > >>> If i am not mistaken and anyone welcome to correct me i just told Edmar > >>> this is set in freeswitch/conf/vars.xml ... file > >>> Ed > >>> > >>> > >>> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz > >>> wrote: > >>> > >>>> > >>>> When I calling an outbound extension it appears: > >>>> > >>>> name is FreeSWITCH and number is 000000000 > >>>> > >>>> How can i change it depends on the user who is calling? > >>>> > >>>> Sample 1001->64521223 > >>>> > >>>> I just want the name 1001 to appear not FreeSWITCH same as the number > >>>> > >>>> Thanks > >>>> > >>>> > >>>> > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24160712.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/481a91d0/attachment.html From anthony.minessale at gmail.com Tue Jun 23 06:29:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 08:29:13 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090623040853.GA84157@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> Message-ID: <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> You are way off base in a few places, let me see if I can clarify a bit. Here are at least 2 pointers: 1) The release tarballs do not come with bootstrap because they already are bootstrapped. 2) FreeSWITCH does not depend on system libs so all the stuff about apr is barking up the wrong tree. we build our own apr and apr-utils I suggest you try latest svn trunk of FS and follow the BSD build guidelines on the WIKI since you say it's closely compatible. On Mon, Jun 22, 2009 at 11:08 PM, Vincent Stemen < vince.freeswitch at hightek.org> wrote: > Hi. > > I have been searching for an alternative PBX to asterisk (which has not > been all that stable) to run on Dragonfly BSD. I spent a fair amount of > time a couple months ago trying to compile freeswitch without success. > > I have since tried Yate, but it consumes 85-95% of the cpu when idle > (not processing any calls). > > I am considering revisiting freeswitch. I have included below my notes > of all the various problems I encountered before, which ones I resolved > and how, up to the point where I left off. I am in hopes of getting > feedback on whether any of the issues have been fixed or are planned to > be fixed and/or suggestions on getting it working. Also, perhaps these > notes on my experiences will be helpful for the developers to improve > freeswitch. > > Keep in mind that Dragonfly is a branch from Freebsd and stays fairly > compatible. I am able to compile most software, that is ported to > FreeBSD, with few problems. > > Here are my notes > ================= > > To compile on dragonfly BSD 1.10.1-RELEASE > ========================================== > I had to add -D__FreeBSD__ to CPPFLAGS > ln sh to bash or zsh > because I got "unexpected operator" errors from "test" during configure > with the bsd shell. > ln make to gmake > Their scripts were calling make even though I ran the build > using gmake. > Must have apr-0.9.16.2.0.61 and apr-util-0.9.16.2.0.61 installed > apr-0.xxx has headers, which freeswitch is including, that apr-1.xxx > does not have. > I sym-linked the apr-util libs from apr-util-1.2.8nb1 because > apr-util-0.9.16.2.0.61 was not available as a binary package. > > The freeswitch-1.0.3.tar.gz release did not have bootstrap.sh, which is > required for building from the svn repository. > > Somebody on the #freeswitch IRC suggested I get it from the subversion > repository and run the bootstrap script. bootstrap.sh is not in the > release. > > ========================================== > Tue Mar 31 00:12:37 CDT 2009 > > I posted on the freeswitch mailing list asking about the compilation errors > Got no responses. > > This first set of apr_... warnings turned out to clearly be from not having > the > correct apr-util package installed, I should have gotten a response on the > list > about it, considering it is a clear dependency that they do not directly > specify on the web site or the source docs. apr-util is a dependency of > subversion. They list SVN as a dependency of freeswitch, which is the > utility > in the subversion package, not the package name. svn should not be > a dependency to build from a release archive that is not retrieved from the > svn > repository. > > As it turns out, it had to be apr-util version 0.9.15. See notes below. > ========================================== > > Tue Mar 31 22:22:58 CDT 2009 > > I tried the freeswitch-snapshot.tar.gz from the freeswitch site, which is > a 03/30/2009 snapshot from the svn trunk. > > *concern* > It was nearly twice as big as the release for some reason. > 27016871 Mar 28 13:08 freeswitch-1.0.3.tar.gz > 52854882 Mar 31 18:25 freeswitch-snapshot.tar.gz > > Running bootstrap.sh produced a bunch of these errors from automake: > > Use of uninitialized value in exists at /usr/pkg/bin/automake line 4823, > line 1. > Use of uninitialized value in concatenation (.) or string at > /usr/pkg/bin/automake line 4823, line 1. > automake: #################### > automake: ## Internal Error ## > automake: #################### > automake: unrequested trace `' > automake: Please contact . > at /usr/pkg/share/automake-1.10/Automake/Channels.pm line 570 > Automake::Channels::msg('automake', '', 'unrequested trace `\'') > called at /usr/pkg/share/automake-1.10/Automake/ChannelDefs.pm line 191 > Automake::ChannelDefs::prog_error('unrequested trace `\'') called at > /usr/pkg/bin/automake line 4823 > Automake::scan_autoconf_traces('configure.ac') called at > /usr/pkg/bin/automake line 5046 > Automake::scan_autoconf_files() called at /usr/pkg/bin/automake line > 781 > > > ========================================== > Thu Apr 2 20:51:44 CDT 2009 > > Going back to the 1.0.3 release. > > I was getting a a bunch of apr_... warnings like > > Compiling src/switch_apr.c ... > src/switch_apr.c: In function `switch_thread_self': > src/switch_apr.c:74: warning: implicit declaration of function > `apr_os_thread_current' > src/switch_apr.c:74: warning: return makes pointer from integer without > a cast > ... > Then > gmake[3]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > > I finally got past that by installing apr-0.9.16.2.0.61 and > apr-util-0.9.15. > I compiled apr-util-0.9.15 myself because it was not available as a binary > package. > Source files are including headers from apr-util-0.9.15 that do not exist > in the > binary package, apr-util-1.2.8nb1.tgz. > > I also got past the "Cannot guess build type" error during configure by > adding > "--build=i386" instead of having to use the uname wrapper to fake FreeBSD. > > ========================================== > New error: (This one is not the fault of freeswitch) > > Compiling src/switch_core.c ... > src/switch_core.c: In function `switch_core_setrlimits': > src/switch_core.c:795: error: `RLIMIT_AS' undeclared (first use in this > function) > > Turns out RLIMIT_AS is defined in /usr/include/sys/resource.h on some other > systems including FreeBSD but not Dragonfly. > > It is used in the source like this > #if !defined(__OpenBSD__) && !defined(__NetBSD__) > setrlimit(RLIMIT_AS, &rlp); > > It is defined in resource.h in FreeBSD like this > #define RLIMIT_AS RLIMIT_VMEM /* standard name for RLIMIT_VMEM > */ > > I created a patch for src/switch_core.c which fixed this error. > > Since then, I discussed this on the Dragonfly BSD mailing list and it is > apparently going to be fixed on a future release of Dragonfly now that > they are aware of it. > > ========================================== > New error: > > Tons of warnings like > warning: return makes pointer from integer without a cast > (this needs cleaned up) > > Then error'd with > gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by ` > libsofia-sip-ua.la'. Stop. > > I cd'd to libs/sofia-sip/libsofia-sip-ua/su > and ran make there. It completed successfully in spite of the same > warnings. > > Then ran make again form the top. > > Got further until > > ========================================== > New error: > > making all mod_spidermonkey > cd config; gmake -j1 export > cd pr; gmake -j1 export > cd include; gmake export > cd md; gmake export > ../../../config/./nsinstall: cannot make symbolic link > /u1/falcon/ports/freeswitch-1.0.3/work/freeswitch-1.0.3/libs/js/nsprpub/dist/include/nspr/.: > File exists > > Why would it be trying to make a sym-link of '.' ?? > > ========================================== > > Giving up for now. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/62794fec/attachment-0001.html From Claudio.Cavalera at italtel.it Tue Jun 23 06:33:02 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Tue, 23 Jun 2009 15:33:02 +0200 Subject: [Freeswitch-users] Variable manipulation in the dialplan Message-ID: Hello, I once found in the wiki a page explaining how to "substring" a channel variable, something like <@[intra]lanman> 12345 would be 345 if you do ${var:2} I can't find that page on the wiki anymore, any hint on were it could be? :-) Also do you think it could be useful to extend this functionality with a sort of Java indexOf() to extract a specific substring from a variable (but without knowing its size like in the example above)? Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From anthony.minessale at gmail.com Tue Jun 23 06:44:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 08:44:43 -0500 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Message-ID: <191c3a030906230644i154a6198j2cf1027e688297d3@mail.gmail.com> try adding this to your jingle in client.xml then edit acl.conf.xml and add this list this tells mod_dingaling that it should only pick candidates that pass the acl list given the one we made called wan excludes all the private ranges. If you update to latest trunk this list is created internally as "wan.auto" so you can use that instead of making one in your config. On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > I am trying to call from my corporate network (firewalled) using Gtalk > to Freeswitch. I am not getting any audio. > > In the logs I see that mod_dingaling is using my internal corporate IP > address which is not publically addressable. > > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates > 146.xx.xx.xx:50320 > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable > Candidate 146.xx.xx.xx:50320 > > Further on in the log, I can see GTalk sending a new candidate IP > address to use: > 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 > name=rtp > type=local > protocol=udp > username=e+JTkVHT1xEkqXGD > password=fAxU6Pr1oF9Zq48U > address=192.168.1.102 > port=50322 > pref=1.00 > > 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked > an IP [146.xx.xx.xx] > > and > > 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 > name=rtp > type=stun > protocol=udp > username=RBqyF2XNMYLfJNoU > password=DQMjon1fSVoJIRTp > address=124.xxx.xxx.xxx > port=50323 > pref=0.90 > > 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked > an IP [146.xx.xx.xx] > and > > 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 > name=rtp > type=relay > protocol=udp > username=62L5zs2FHbcUdeCJ > password=KxmNgkUmZsLfuX6S > address=209.xx.xxx.xxx > port=19295 > pref=0.50 > > 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked > an IP [146.xx.xx.xx] > > Because of this, I never get audio. Any ideas how to fix this? > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/dbf5783c/attachment.html From anthony.minessale at gmail.com Tue Jun 23 06:49:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 08:49:23 -0500 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: References: Message-ID: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> play with it from the cli freeswitch>global_setvar foo=12345 API CALL [global_setvar(foo=12345)] output: +OK freeswitch> eval ${foo:2:1} API CALL [eval(${foo:2:1})] output: 3 freeswitch> eval ${foo:2:3} API CALL [eval(${foo:2:3})] output: 345 freeswitch> eval ${foo:3:2} API CALL [eval(${foo:3:2})] output: 45 freeswitch> eval ${foo:-4:4} API CALL [eval(${foo:-4:4})] output: 2345 On Tue, Jun 23, 2009 at 8:33 AM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > Hello, > I once found in the wiki a page explaining how to "substring" a channel > variable, > something like > <@[intra]lanman> 12345 would be 345 if you do ${var:2} > > I can't find that page on the wiki anymore, any hint on were it could > be? :-) > > Also do you think it could be useful to extend this functionality with a > sort of Java indexOf() to extract a specific substring from a variable > (but without knowing its size like in the example above)? > > Regards, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/237e5c1a/attachment.html From brian at freeswitch.org Tue Jun 23 06:23:31 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 08:23:31 -0500 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Message-ID: <41DAEAFC-5EF1-4637-8CBE-92CDC9F83E72@freeswitch.org> No it snot because of this.. you have to understand how Jingle works and if you notice it has three candidates.... 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Its already picked this one, maybe a packet capture would clear this up. /b On Jun 23, 2009, at 7:51 AM, Mark Campbell-Smith wrote: > Because of this, I never get audio. Any ideas how to fix this? > > Thanks! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/2b8f1d3c/attachment-0001.html From max.bridgewater at gmail.com Tue Jun 23 07:04:25 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 23 Jun 2009 10:04:25 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> References: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> Message-ID: Hi Michael, Using loopback solves my problem. Thanks a lot. There is a strange thing i observed though. I need to paste my extension in the default.xml file. Having them in the default directory isn't enough. Is that normal? Max. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/307fd05f/attachment.html From msc at freeswitch.org Tue Jun 23 07:44:51 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 23 Jun 2009 07:44:51 -0700 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> Message-ID: <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> On Jun 23, 2009, at 7:04 AM, Max Bridgewater wrote: > > Hi Michael, > > Using loopback solves my problem. Thanks a lot. > There is a strange thing i observed though. I need to paste my > extension in the default.xml file. Having them in the default > directory isn't enough. Is that normal? > No it isn't. What is the name of the file that has your extension and what subdir is it in? Can you pb the contents? -MC > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From max.bridgewater at gmail.com Tue Jun 23 07:59:56 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 23 Jun 2009 10:59:56 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> References: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> Message-ID: The file is located under /usr/local/freeswitch/conf/dialplan/default/. The name is: mysocket.xml. The content is: Max. On Tue, Jun 23, 2009 at 10:44 AM, Michael S Collins wrote: > > On Jun 23, 2009, at 7:04 AM, Max Bridgewater > wrote: > > > > > Hi Michael, > > > > Using loopback solves my problem. Thanks a lot. > > There is a strange thing i observed though. I need to paste my > > extension in the default.xml file. Having them in the default > > directory isn't enough. Is that normal? > > > > No it isn't. What is the name of the file that has your extension and > what subdir is it in? Can you pb the contents? > -MC > > > Max. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/52e11a9b/attachment.html From mike at jerris.com Tue Jun 23 08:24:59 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 11:24:59 -0400 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia In-Reply-To: References: Message-ID: If you can supply a patch to expose this as a config option for us it would be appreciated. Patches can be posted to http://jira.freeswitch.org . Mike On Jun 17, 2009, at 3:22 PM, Muhammad Shahzad wrote: > Ok, thanks, i will take care of it in my code where necessary. > > Thank you. > > > On Thu, Jun 18, 2009 at 12:54 AM, Brian West > wrote: > Its not possible right now but you could if you enable the config > option and apply the tag... its something I have thought about adding > but wasn't high on my list. > > NTATAG_SIPFLAGS(MSG_FLG_COMPACT) > > http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 > > /b > > On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: > > > Hi, > > > > Is it possible to enable compact SIP headers in mod_sofia > > configuration? If yes, then how to do so? Kindly give an example. > > > > Thank you. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/0e128c86/attachment.html From rupa at rupa.com Tue Jun 23 08:38:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 23 Jun 2009 10:38:51 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <005401c9f406$3ced3710$b6c7a530$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> Message-ID: How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg > and mac/phone specific cfg? settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns > *Sent:* Monday, June 22, 2009 2:57 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Sounds like a config issue in the tag. Check global sip.cfg and > mac/phone specific cfg. When you are dialing on-hook I don't think it will > use your .digitmap or ..digitmap.timer settings. When you dial off-hook it > sure will. > > On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines > on the phone. The first two are registered with a SwitchVox, the last with > Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6ae9eac9/attachment-0001.html From apt.get at gmail.com Tue Jun 23 09:01:30 2009 From: apt.get at gmail.com (David Burgess) Date: Tue, 23 Jun 2009 10:01:30 -0600 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <4A404CAA.5080809@isp-systems.net> References: <4A404CAA.5080809@isp-systems.net> Message-ID: On Mon, Jun 22, 2009 at 9:31 PM, murrah boswell wrote: > Hello All, > > I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance > on how to setup a testbed in a thin client environment. I think this would be a fairly simple matter of installing freeswitch and your softphone of choice on your ltsp server and configuring your extensions to register on 127.0.0.1, or whatever interface your freeswitch internal profile will be active on. In other words, ltsp is designed such that you can install your telephony on the server and the ltsp infrastructure will proliferate that functionality to your thin client. Install freeswitch and ekiga on the server and get ekiga to register. If you have trouble with that then this would be the place to ask. Once you get your ekiga extension registered and you are able to call voice mail, moh, etc, then log into a thin client and try the same from there. I think it will just work, but if not, that would be a good problem for the ltsp-discuss mailing list. https://lists.sourceforge.net/lists/listinfo/ltsp-discuss db From mike at jerris.com Tue Jun 23 09:11:16 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:11:16 -0400 Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <24109532.post@talk.nabble.com> References: <24109532.post@talk.nabble.com> Message-ID: <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> Try turning up your logging level to debug to see why the call is hanging up. Mike On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: > > My freeswitch has a mysql database consists of freeswitch tables, > registrations and nibblebill on mysql configured it correctly and > working... > Issue is when I call external ip's sometimes it works sometimes not? > > 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 > switch_core_session_enable_heartbeat() sofia/internal/ > 1006 at 116.5.231.40 > setting session heartbeat to 1 second(s). > 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 > switch_core_standard_on_execute() Hangup sofia/internal/1006 at 116.50.231.72 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40 > ) > Ended > 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/internal/1006 at 116.5.231.40 > [CS_DESTROY] > > On my acl.conf.xml I allow ip 116.5.231.40 > > > > > > > > I put this on my external and internal profile > > param name="apply-inbound-acl" value="globals"/> > > And put auth-calls to false... > > Please help me am really near to my success here in freeswitch... > Thanks... From mike at jerris.com Tue Jun 23 09:20:06 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:20:06 -0400 Subject: [Freeswitch-users] channel variable sip_to_tag In-Reply-To: <4A3E0607.1060608@xpirio.com> References: <4A3E0607.1060608@xpirio.com> Message-ID: if you need to use the same tags, we should be using the whole same nh in the code. There is code to do this by call uuid but I can't recall if thats for NOTIFY or INFO. If its the wrong one, we should add teh same for what you need. Mike On Jun 21, 2009, at 6:05 AM, Christian L?schenkohl wrote: > hello > > do someone know how to get the sip_to_tag from an active call? > the sip_from_tag is available as a channel variable but sip_to_tag > isn't. > i don't know if it is available at call setup, the fist time i see > the tag=... > in the sip header is the challenge response answer from fs > > i need this to get my aoc (advice-of-charge) implementation running, > this one > is based on sip info messages and has to contain the same tag's as > the active call. > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 23 09:26:30 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:26:30 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: <004e01c9f378$c1707e90$44517bb0$@net> References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> <004e01c9f378$c1707e90$44517bb0$@net> Message-ID: <1257CF28-EE67-4E9F-8B1A-1A25F1A7C27A@jerris.com> if you run from the visual studio ide, there is a windows that shows the stack trace. Mike On Jun 22, 2009, at 4:33 PM, Drago Totev wrote: > Michael, > > It is definitely a problem in this build (13754M) (Windows) > > I found build from March and it does not brakes (still no joy, but > not sure for now if the exchange side is configured properly). > > Again, I would submit bug report is you can point me to instructions > how to do collect backtrace on Windows OS. > > Drago > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Jerris > Sent: Sunday, June 21, 2009 4:31 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 > > If this is still the case with current svn trunk, please get a > backtrace and post it to http://jira.freeswitch.org > > On Jun 21, 2009, at 3:51 PM, "Drago Totev" > wrote: > After days of running around clueless, I have no other option but to > ask the community for help one last time? > > Here is what happens: > > 1. FS sends INVITE > > 2. Exchange answers with ?302 Moved Temporarily? > > 3. FS bombs and closes > > > > Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/ > 2.0 > Message Header > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > Max-Forwards: 68 > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > To: > SIP to address: sip:4783874764 at 10.0.0.71 > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > CSeq: 116688296 INVITE > Contact: > Contact Binding: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Privacy: none > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 236 > P-Asserted-Identity: "MILLEDGEVL GA" > > Message Body > > > > Status-Line: SIP/2.0 100 Trying > Message Header > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > TO: > SIP to address: sip:4783874764 at 10.0.0.71 > CSEQ: 116688296 INVITE > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > CONTENT-LENGTH: 0 > > > Status-Line: SIP/2.0 302 Moved Temporarily > Message Header > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > TO: ;tag=27df6afe0 > SIP to address: sip:4783874764 at 10.0.0.71 > SIP tag: 27df6afe0 > CSEQ: 116688296 INVITE > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > CONTACT: > Contact Binding: > > CONTENT-LENGTH: 0 > SERVER: RTCC/3.0.0.0 > > > Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 > Message Header > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > Max-Forwards: 68 > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > To: ;tag=27df6afe0 > SIP to address: sip:4783874764 at 10.0.0.71 > SIP tag: 27df6afe0 > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > CSeq: 116688296 ACK > Content-Length: 0 > > > Here is the FS log beginning the the processing of the call: > > 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 > [637f8a72-8034-254f-9937-b321c0d87414] > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 > entering state [received][100] > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: > v=0 > o=NXT02 19785 8060 IN IP4 209.249.3.59 > s=sip call > c=IN IP4 209.249.3.60 > t=0 0 > m=audio 36292 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec > Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 > PCMU/8000 20 ms 160 samples > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf > payload to 101 > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_NEW > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59 > ) State NEW > 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_NEW -> CS_INIT > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_INIT > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59 > ) State INIT > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 > SOFIA INIT > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_INIT -> CS_ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59 > ) State INIT going to sleep > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59 > ) State ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 > SOFIA ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 > Standard ROUTING > 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing > MILLEDGEVL GA->4783874190 in context public > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >unloop] continue=false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >outside_call] continue=true > Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition > [outside_call] > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > set(outside_call=true) > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >call_debug] continue=true > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [call_debug] ${call_debug}(true) =~ /^true$/ break=never > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >public_extensions] continue=false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9]) > $/ break=on-false > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >Local_UM] continue=false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on- > false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_ROUTING -> CS_EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59 > ) State ROUTING going to sleep > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59 > ) State EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 > SOFIA EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 > Standard EXECUTE > EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) > 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 > SET [outside_call]=[true] > EXECUTE sofia/external/4782512197 at 209.249.3.59 info() > 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Channel-Read-Codec-Name: [PCMU] > Channel-Read-Codec-Rate: [8000] > Channel-Write-Codec-Name: [PCMU] > Channel-Write-Codec-Rate: [8000] > Caller-Username: [4782512197] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [MILLEDGEVL GA] > Caller-Caller-ID-Number: [4782512197] > Caller-Network-Addr: [209.249.3.59] > Caller-Destination-Number: [4783874190] > Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > Caller-Source: [mod_sofia] > Caller-Context: [public] > Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1245613758636018] > Caller-Channel-Created-Time: [1245613758636018] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [209.249.3.59] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [4782512197] > variable_sip_from_uri: [4782512197 at 209.249.3.59] > variable_sip_from_host: [209.249.3.59] > variable_sip_from_user_stripped: [4782512197] > variable_sip_from_tag: [3454602382-411732] > variable_sofia_profile_name: [external] > variable_sip_cid_type: [pid] > variable_sip_req_user: [gw+Broadvox] > variable_sip_req_port: [5080] > variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] > variable_sip_req_host: [71.29.0.61] > variable_sip_to_user: [4783874764] > variable_sip_to_port: [5060] > variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] > variable_sip_to_host: [209.249.3.56] > variable_sip_contact_user: [4782512197] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] > variable_sip_contact_host: [209.249.3.59] > variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] > variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] > variable_sip_via_host: [209.249.3.59] > variable_sip_via_port: [5060] > variable_max_forwards: [69] > variable_sip_call_info: [ 209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"] > variable_sip_gateway: [Broadvox] > variable_switch_r_sdp: [v=0 > o=NXT02 19785 8060 IN IP4 209.249.3.59 > s=sip call > c=IN IP4 209.249.3.60 > t=0 0 > m=audio 36292 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ] > variable_remote_media_ip: [209.249.3.60] > variable_remote_media_port: [36292] > variable_read_codec: [PCMU] > variable_read_rate: [8000] > variable_write_codec: [PCMU] > variable_write_rate: [8000] > variable_endpoint_disposition: [RECEIVED] > variable_outside_call: [true] > variable_current_application: [info] > > > EXECUTE sofia/external/4782512197 at 209.249.3.59 > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 > variable string 0 = [absolute_codec_string=PCMA] > 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] > 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/ > 4783874764) State Change CS_NEW -> CS_INIT > 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_INIT > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/ > 4783874764 SOFIA INIT > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 4783874764) State Change CS_INIT -> CS_ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT going to sleep > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/ > 4783874764 SOFIA ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/ > internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING going to sleep > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 > (sofia/internal/4783874764) State CONSUME_MEDIA > 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/ > internal/4783874764 entering state [calling][0] > > > Will trade my first born for little help J > > Drago > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Drago Totev > Sent: Sunday, June 14, 2009 7:11 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 > > Hello everyone, > > I am trying to set a test environment with FS and Exchange 2007. > > Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM > and it does not seem to work. Thus far I use the default settings, > one ATA registered and confirmed to work. > > According external source: > > ?MS Exchange does not understand 'From' fields with domain name in > the address (example: field containing > address).The ?From? field?s address to be in the format ? >?; otherwise, MS Exchange drops the call.? > > I don?t know if this is the only problem? However, I see exactly > this behavior: > > ?PBX initiates call to MS Exchange by sending a SIP INVITE message > to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved > Temporarily) response to PBX asking to repeat INVITE to a different > port (5065 for example). After PBX repeats the INVITE sending, the > call is established.? > > After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without > any record in the log. > > Can someone help with working configuration, please? Exchange 2007 > UM role Version: 08.01.0359.002 > > Thanks in advance. > > Drago > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/9056a56b/attachment-0001.html From Richard.Lamkin at mettoni.com Tue Jun 23 09:31:06 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 23 Jun 2009 17:31:06 +0100 Subject: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. 2 - I do not want that incoming call to be answered but just stay ringing. 3 - Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. So far I've managed 1 - I see on the incoming call on the event API 2 - I used sleep 180000 (3 mins) see rule below. 3 - failed - because the rule is executing a sleep command and I cannot break in with my redirect. ============ I have tested the following works as single DP rule. Using the fixed dial plan rule below I do get the SIP signalling I want but of course it's a redirect immediately and to a fixed destination. The redirect causes FS to send a "302 moved temporarily", and the move works. ============= Any suggestions would be gratefully received Richard Lamkin richard.lamkin at mettoni.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/802f759d/attachment.html From mike at jerris.com Tue Jun 23 09:36:31 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:36:31 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED In-Reply-To: <24158819.post@talk.nabble.com> References: <24143545.post@talk.nabble.com> <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> <24158819.post@talk.nabble.com> Message-ID: if you turn up the debug logs it should tell you why. On Jun 22, 2009, at 11:38 PM, Edmar Cruz wrote: > > Nope. I just want to call a mobile number with no register number. > > Brian West-3 wrote: >> >> I'm going to guess you're calling a registered user? If so replace >> the @ with % >> >> /b >> >> On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: >> >>> >>> Hi, >>> >>> API CALL [originate sofia/external/1001 at 116.50.456.212] >>> -ERR SERVICE_NOT_IMPLEMENTED >>> >>> I receiving this error i dont know y? Can u help mo on this? >>> >>> I dialing a mobile number on this sometimes it works... Sometimes it >>> destroys the call [CALL_DESTROY] >>> From brian at freeswitch.org Tue Jun 23 09:36:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 11:36:56 -0500 Subject: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> Message-ID: On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote: > Can anyone suggest a good way to do the following; > > 1 - I want to be alerted [via the event API] to a new incoming call. See below.. ie park. You should get an event via event socket you can decide what to do. > 2 - I do not want that incoming call to be answered but just stay > ringing. Can't really do it that way.. you can answer it but then you're responsible for generating ringback. And billing starts when you answer it. > 3 ? Then via the API I want to send a redirect command to push the > call off to a new destination of my choice, I do not want to use the > answer/deflect sequence. Try using park ... this way you put the call in limbo and you can send the call commands at your leisure. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park > So far I?ve managed > 1 - I see on the incoming call on the event API > 2 ? I used sleep 180000 (3 mins) see rule below. > 3 ? failed - because the rule is executing a sleep command and I > cannot break in with my redirect. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/17d60d6e/attachment.html From chris at cloudtel.com Tue Jun 23 09:36:19 2009 From: chris at cloudtel.com (Chris Burns) Date: Tue, 23 Jun 2009 11:36:19 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> Message-ID: Basically read the polycom manual ... it is the polycom producing the dialtone and deciding when to dial the number you are entering, using its own dialplan and interdigit timers. On Tue, Jun 23, 2009 at 10:38 AM, Rupa Schomaker wrote: > How are you configuring your polycom? > > > On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > >> I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg >> and mac/phone specific cfg? settings. I also looked for digitmap but could >> find nothing. >> >> >> >> Can you be more specific? >> >> >> >> Thanks, Lars >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns >> *Sent:* Monday, June 22, 2009 2:57 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Polycom configuration problems? >> >> >> >> Sounds like a config issue in the tag. Check global sip.cfg >> and mac/phone specific cfg. When you are dialing on-hook I don't think it >> will use your .digitmap or ..digitmap.timer settings. When you dial off-hook >> it sure will. >> >> On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: >> >> I am having difficulty with a Polycom 501 and Freeswitch. There are 3 >> lines on the phone. The first two are registered with a SwitchVox, the last >> with Freeswitch. >> >> >> >> When I select the 3rd line and begin to press numbers, pressing the 3rd >> digit automatically causes the phone to begin to dial. It does not matter >> which three numbers I press, the 3rd one is magic. >> >> >> >> However, if I do not select a line before dialing and key a 10-digit >> number into the phone, then select the 3rd line, it dials out fine. >> >> >> >> You can see from the debug console output that Processing begins before it >> hits any dialplan, so that cannot be the problem. I must have the line >> defined incorrectly for Freeswitch. >> >> >> >> Thanks for any suggestions, Lars. >> >> >> >> PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 >> >> >> >> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 >> i386 GNU/Linux >> >> >> >> 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected >> by acl "domains". Falling back to Digest auth. >> >> 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected >> by acl "domains". Falling back to Digest auth. >> >> 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel >> sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] >> >> 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 >> (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ >> 1001 at 192.168.10.29 entering state [received][100] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: >> >> v=0 >> >> o=- 1245682011 1245682011 IN IP4 192.168.10.101 >> >> s=Polycom IP Phone >> >> c=IN IP4 192.168.10.101 >> >> t=0 0 >> >> m=audio 2254 RTP/AVP 0 8 18 101 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:18 G729/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> >> >> 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 >> (sofia/internal/1001 at 192.168.10.29) State NEW >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[G7221:115:32000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[G7221:107:16000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[G722:9:8000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[PCMU:0:8000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec >> sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload >> to 101 >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ >> 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT >> >> 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1001 at 192.168.10.29 [BREAK] >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 >> (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 >> (sofia/internal/1001 at 192.168.10.29) State INIT >> >> 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ >> 1001 at 192.168.10.29 SOFIA INIT >> >> 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ >> 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1001 at 192.168.10.29 [BREAK] >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 >> (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 >> (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 >> (sofia/internal/1001 at 192.168.10.29) State ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ >> 1001 at 192.168.10.29 SOFIA ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 >> sofia/internal/1001 at 192.168.10.29 Standard ROUTING >> >> 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing >> 1001->323 in context default >> >> Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] >> continue=false >> >> Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> >> Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/942f5c09/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Jun 23 09:45:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 23 Jun 2009 17:45:39 +0100 Subject: [Freeswitch-users] Sound file or lua script not played under load Message-ID: Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/05fcd011/attachment.html From nik.middleton at noblesolutions.co.uk Tue Jun 23 09:54:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 23 Jun 2009 17:54:34 +0100 Subject: [Freeswitch-users] Sound file or lua script not played under load In-Reply-To: References: Message-ID: Hmm, Looking at console I'm seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 23 June 2009 17:46 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/f7359da4/attachment.html From mattdfong at gmail.com Tue Jun 23 09:55:11 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 23 Jun 2009 09:55:11 -0700 Subject: [Freeswitch-users] Sound file or lua script not played under load In-Reply-To: References: Message-ID: <4256bf830906230955s60487187t350808c0c6075e7b@mail.gmail.com> Does the log show anything? if the lua script fails to execute it should appear in freeswitch.log On Tue, Jun 23, 2009 at 9:45 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Scratching my head on this one, under load FS is not playing an audio file, > OR and lua script is not getting executed. Not all the time, just some. > I?ve changed ulimit ?n to 99999 but no diff, and ideas where else I might > look? > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/8ed2a1dd/attachment.html From larclap at yahoo.com Tue Jun 23 10:25:32 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 23 Jun 2009 10:25:32 -0700 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> Message-ID: <00f301c9f427$9cd311b0$d6793510$@com> Via a web browser. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: I'm sorry Chris, but I don't know where the look for the "global sip.cfg and mac/phone specific cfg" settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/58aa28ba/attachment-0001.html From rupa at rupa.com Tue Jun 23 10:46:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 23 Jun 2009 12:46:14 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <00f301c9f427$9cd311b0$d6793510$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> Message-ID: Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: > Via a web browser. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 23, 2009 8:39 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > How are you configuring your polycom? > > On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > > I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg > and mac/phone specific cfg? settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns > *Sent:* Monday, June 22, 2009 2:57 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Sounds like a config issue in the tag. Check global sip.cfg and > mac/phone specific cfg. When you are dialing on-hook I don't think it will > use your .digitmap or ...digitmap.timer settings. When you dial off-hook it > sure will. > > On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines > on the phone. The first two are registered with a SwitchVox, the last with > Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/3b186550/attachment.html From anthony.minessale at gmail.com Tue Jun 23 11:21:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 13:21:07 -0500 Subject: [Freeswitch-users] Sound file or lua script not played under load In-Reply-To: References: Message-ID: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hmm, > > > > Looking at console I?m seeing this, does this offer any additional clues to > anyone? > > > > 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nik > Middleton > *Sent:* 23 June 2009 17:46 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Sound file or lua script not played under > load > > > > Hi Guys, > > > > Scratching my head on this one, under load FS is not playing an audio file, > OR and lua script is not getting executed. Not all the time, just some. > I?ve changed ulimit ?n to 99999 but no diff, and ideas where else I might > look? > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/c0b083d2/attachment-0001.html From christian.loeschenkohl at xpirio.com Tue Jun 23 11:27:19 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Tue, 23 Jun 2009 20:27:19 +0200 Subject: [Freeswitch-users] channel variable sip_to_tag In-Reply-To: References: <4A3E0607.1060608@xpirio.com> Message-ID: <4A411E87.6000805@xpirio.com> hi thank you for your reply how can we procced? br On 2009-06-23 18:20, Michael Jerris wrote: > if you need to use the same tags, we should be using the whole same nh > in the code. There is code to do this by call uuid but I can't recall > if thats for NOTIFY or INFO. If its the wrong one, we should add teh > same for what you need. > > Mike > > On Jun 21, 2009, at 6:05 AM, Christian L?schenkohl wrote: > >> hello >> >> do someone know how to get the sip_to_tag from an active call? >> the sip_from_tag is available as a channel variable but sip_to_tag >> isn't. >> i don't know if it is available at call setup, the fist time i see >> the tag=... >> in the sip header is the challenge response answer from fs >> >> i need this to get my aoc (advice-of-charge) implementation running, >> this one >> is based on sip info messages and has to contain the same tag's as >> the active call. >> >> br >> >> -- >> Ing. Christian L?schenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl at xpirio.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From nik.middleton at noblesolutions.co.uk Tue Jun 23 11:35:16 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 23 Jun 2009 19:35:16 +0100 Subject: [Freeswitch-users] Sound file or lua script not played underload In-Reply-To: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> References: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> Message-ID: They're reading an audio file from a ram disk. Wouldn't have thought that this would cause a problem or am I wrong. Running at around 400 concurrent calls Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 23 June 2009 19:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sound file or lua script not played underload Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton wrote: Hmm, Looking at console I'm seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 23 June 2009 17:46 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/ac6a431e/attachment.html From anthony.minessale at gmail.com Tue Jun 23 11:48:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 13:48:04 -0500 Subject: [Freeswitch-users] Sound file or lua script not played underload In-Reply-To: References: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> Message-ID: <191c3a030906231148t771f590csa3cd8c46c5432b5@mail.gmail.com> the lines you pasted indicate something stuck playing local_stream (hold music) and not actually reading it. playing a file from a ram disk with 400 is for sure fine. I have done many thousand before. if you turn up your debugging do you see anything else about the box going wrong? On Tue, Jun 23, 2009 at 1:35 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > They?re reading an audio file from a ram disk. Wouldn?t have thought > that this would cause a problem or am I wrong. Running at around 400 > concurrent calls > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 23 June 2009 19:21 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sound file or lua script not played > underload > > > > Are you making many calls share a single local_stream? > This error usually means a handle open to a local_stream is not reading > from that stream source, such as if you paused during playback of a > local_stream. > They are only a real issue if you are getting them with no calls up. > > On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hmm, > > > > Looking at console I?m seeing this, does this offer any additional clues to > anyone? > > > > 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nik > Middleton > *Sent:* 23 June 2009 17:46 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Sound file or lua script not played under > load > > > > Hi Guys, > > > > Scratching my head on this one, under load FS is not playing an audio file, > OR and lua script is not getting executed. Not all the time, just some. > I?ve changed ulimit ?n to 99999 but no diff, and ideas where else I might > look? > > > > Regards, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/31b8ef56/attachment-0001.html From max.bridgewater at gmail.com Tue Jun 23 12:11:52 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 23 Jun 2009 15:11:52 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> Message-ID: Hi, I've got some news on this. When i move my extension to a different directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include element at the very sample place where the default is included, things work just as expected. That is, my default.xml now include following: Cheers, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/cf95c1cb/attachment.html From msc at freeswitch.org Tue Jun 23 12:27:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 12:27:26 -0700 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> Message-ID: <87f2f3b90906231227s6070f941q97f37ecdb8a2e43f@mail.gmail.com> I love it when users figure it out AND report back what they did to solve the issue! Nice work. -MC On Tue, Jun 23, 2009 at 12:11 PM, Max Bridgewater wrote: > Hi, > > I've got some news on this. When i move my extension to a different > directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include > element at the very sample place where the default is included, things work > just as expected. That is, my default.xml now include following: > > > > > Cheers, > Max. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/a5ac44cf/attachment.html From msc at freeswitch.org Tue Jun 23 12:35:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 12:35:50 -0700 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <191c3a030906230644i154a6198j2cf1027e688297d3@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> <191c3a030906230644i154a6198j2cf1027e688297d3@mail.gmail.com> Message-ID: <87f2f3b90906231235t382df4c5jed4c95941a68c450@mail.gmail.com> Also, if and when you get this working please send a message to the list. I'd like to make sure that your setup gets documented on the wiki. -MC On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try adding this to your jingle in client.xml > > > > then edit acl.conf.xml and add this list > > > > > > > > this tells mod_dingaling that it should only pick candidates that pass the > acl list given > the one we made called wan excludes all the private ranges. > > If you update to latest trunk this list is created internally as "wan.auto" > so you can use that > instead of making one in your config. > > > > On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith < > mcampbellsmith at gmail.com> wrote: > >> Hi! >> >> I am trying to call from my corporate network (firewalled) using Gtalk >> to Freeswitch. I am not getting any audio. >> >> In the logs I see that mod_dingaling is using my internal corporate IP >> address which is not publically addressable. >> >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates >> 146.xx.xx.xx:50320 >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable >> Candidate 146.xx.xx.xx:50320 >> >> Further on in the log, I can see GTalk sending a new candidate IP >> address to use: >> 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 >> name=rtp >> type=local >> protocol=udp >> username=e+JTkVHT1xEkqXGD >> password=fAxU6Pr1oF9Zq48U >> address=192.168.1.102 >> port=50322 >> pref=1.00 >> >> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked >> an IP [146.xx.xx.xx] >> >> and >> >> 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 >> name=rtp >> type=stun >> protocol=udp >> username=RBqyF2XNMYLfJNoU >> password=DQMjon1fSVoJIRTp >> address=124.xxx.xxx.xxx >> port=50323 >> pref=0.90 >> >> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked >> an IP [146.xx.xx.xx] >> and >> >> 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 >> name=rtp >> type=relay >> protocol=udp >> username=62L5zs2FHbcUdeCJ >> password=KxmNgkUmZsLfuX6S >> address=209.xx.xxx.xxx >> port=19295 >> pref=0.50 >> >> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked >> an IP [146.xx.xx.xx] >> >> Because of this, I never get audio. Any ideas how to fix this? >> >> Thanks! >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/ebf1a027/attachment.html From marketing at cluecon.com Tue Jun 23 12:19:12 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 23 Jun 2009 12:19:12 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Important Information Message-ID: <87f2f3b90906231219le7753c9p3a2940ef9b5ccb58@mail.gmail.com> I know you are all eagerly anticipating the arrival of the coolest conference around! We want to make sure that everyone is aware of the following information: * The last day to get the early-bird registration is Wednesday, July 1. Early birds get into the conference for only $499. After July 1 the price is $699 per person. Please call 877.742.CLUE and get registered today! * The last day to book a hotel room at the Wyndham is Tuesday, July 21. Be sure to use expedia.com to get the best deal available. The ClueCon team is working hard to make this a very special event and we hope to have more announcements soon. You don't want to miss ClueCon 2009 - it will be the best conference you attend this year, bar none! -The ClueCon Team http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/fe040cf9/attachment.html From msc at freeswitch.org Tue Jun 23 12:55:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 12:55:07 -0700 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <4A404CAA.5080809@isp-systems.net> References: <4A404CAA.5080809@isp-systems.net> Message-ID: <87f2f3b90906231255iecb6d4fm4635290051f9cf23@mail.gmail.com> Curious - what kinds of SIP phones do the clients support? Have you decided what you'd be using? -MC On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell wrote: > Hello All, > > I am an absolute newbee in the voip world but have a project where I > believe freeswitch will work and need very, very basic guidance > on how to setup a testbed in a thin client environment. > > I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and > utilize fl_teachertool 0.07 to monitor the connected > terminal clients (TCs). If you are not familiar with fl_teachertool, it > allows a teacher to view thumbnail images of each TC logged > in to the server. The teacher can click on any thumbnail and enlarge the > view, monitor all applications running on a given TC, and > take control of the keyboard and mouse of the TC. These are just a few of > the capabilities of fl_teachertool. > > What I want to do is allow the teacher to establish voice communication > using headsets and microphones with any one of the TCs by > making a "phone" call via ethernet based upon ip of the TC through > freeswitch using a softphone. > > Does this sound like something that is possible using freeswitch? If so, > could someone please give me very basic instructions on how > to setup this proof of concept? If I can just get a "teacher" stationed at > my server talking to one "student" at a TC, I believe I > can go from there. Currently I have a voiper softphone that functions, I > believe, under gnome, but I have no idea how to configure > the voiper to initiate calls through freeswitch or how to configure > freeswitch to route the call to one of my TCs. > > I also need to keep this system fully self contained. That is, I can not > have a requirement to use an outside sip service provider. > > Also, I would use any other linux sip softphones known to work with > freeswitch that people feel would work better than a voiper. > voiper seems to be more windows and mac based. I would really like to use > an ekiga since they seem to be more linux based, but I do > not believe that they have been thoroughly tested with freeswitch. > > Any help would be greatly appreciated! > > > Regards, > Murrah Boswell > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/f1b37ba1/attachment-0001.html From msc at freeswitch.org Tue Jun 23 13:15:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 13:15:31 -0700 Subject: [Freeswitch-users] voicemail problem In-Reply-To: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> References: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> Message-ID: <87f2f3b90906231315s7e8ea782j821997fb46ba7937@mail.gmail.com> Did you ever get resolution on this? If not, join us on IRC and we'll discuss it. -MC On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > I have a problem with voicemail in that freeswitch fails to let users > leave their message. Something wrong in the config I guess. I see > this in the logs: > > 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message] (en:en) > 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU > 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-goodbye.wav] (en:en) > > I assume the vm-record_message.PCMU is the file that will be created > to record the voicemail. Is that correct and how can I fix this? > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/4c014b04/attachment.html From brian at freeswitch.org Tue Jun 23 13:50:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 15:50:17 -0500 Subject: [Freeswitch-users] voicemail problem In-Reply-To: <87f2f3b90906231315s7e8ea782j821997fb46ba7937@mail.gmail.com> References: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> <87f2f3b90906231315s7e8ea782j821997fb46ba7937@mail.gmail.com> Message-ID: You're using native files and you have no native files in PCMU... /b On Jun 23, 2009, at 3:15 PM, Michael Collins wrote: > Did you ever get resolution on this? If not, join us on IRC and > we'll discuss it. > -MC > > On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith > wrote: > Hi! > > I have a problem with voicemail in that freeswitch fails to let users > leave their message. Something wrong in the config I guess. I see > this in the logs: > > 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message] (en:en) > 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm- > record_message.PCMU > 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-goodbye.wav] (en:en) > > I assume the vm-record_message.PCMU is the file that will be created > to record the voicemail. Is that correct and how can I fix this? > > Thanks! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/3fc78964/attachment.html From larclap at yahoo.com Tue Jun 23 14:57:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 23 Jun 2009 14:57:11 -0700 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> Message-ID: <017201c9f44d$90287740$b07965c0$@com> Thanks to Rupa and Chris for this help. I didn't know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out. Are Polycoms the only SIP phones which have this feature? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 10:46 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: Via a web browser. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: I'm sorry Chris, but I don't know where the look for the "global sip.cfg and mac/phone specific cfg" settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ...digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/617fc764/attachment-0001.html From brian at freeswitch.org Tue Jun 23 15:04:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 17:04:34 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <017201c9f44d$90287740$b07965c0$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> <017201c9f44d$90287740$b07965c0$@com> Message-ID: Nope other phones have this also. /b On Jun 23, 2009, at 4:57 PM, Lars Zeb wrote: > Thanks to Rupa and Chris for this help. I didn?t know enough to > understand Chris was pointing me to the Polycom phone rather than > FS. I would never have figured this out. > > Are Polycoms the only SIP phones which have this feature? > > Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/9d1b1f43/attachment.html From rupa at rupa.com Tue Jun 23 15:06:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 23 Jun 2009 17:06:03 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <017201c9f44d$90287740$b07965c0$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> <017201c9f44d$90287740$b07965c0$@com> Message-ID: Every sip phone I've used has this feature. Even ATAs -- though they tend to ship with more forgiving defaults. On Tue, Jun 23, 2009 at 4:57 PM, Lars Zeb wrote: > Thanks to Rupa and Chris for this help. I didn?t know enough to > understand Chris was pointing me to the Polycom phone rather than FS. I > would never have figured this out. > > > > Are Polycoms the only SIP phones which have this feature? > > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 23, 2009 10:46 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Ok, most of us configure the polycoms via a provisioning interface. > usually ftp or http. > > Anyway, when using the web interface, you want to look at: > > Goto the web interface, Click on SIP. > > Scroll down to the Local Settings section and you need to modify digitmap > and digitmap timeout. the syntax is in the polycom manuals which you can > donwload from polycom. > > On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: > > Via a web browser. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 23, 2009 8:39 AM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > How are you configuring your polycom? > > On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > > I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg > and mac/phone specific cfg? settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns > *Sent:* Monday, June 22, 2009 2:57 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Sounds like a config issue in the tag. Check global sip.cfg and > mac/phone specific cfg. When you are dialing on-hook I don't think it will > use your .digitmap or ....digitmap.timer settings. When you dial off-hook it > sure will. > > On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines > on the phone. The first two are registered with a SwitchVox, the last with > Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/c88e83d5/attachment-0001.html From timb0311 at hotmail.com Tue Jun 23 15:12:38 2009 From: timb0311 at hotmail.com (Tim B) Date: Tue, 23 Jun 2009 18:12:38 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: Did anyone have any suggestions on this? Just to reiterate... - 8000 is a local extension defined in the default dialplan... see http://pastebin.freeswitch.org/9450 for definition - didn't work: originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) ... see http://pastebin.freeswitch.org/9440 for log - had to add the FS ip (192.168.10.35) to the domains acl... now it to works Is this the proper way to configure? Tim From: timb0311 at hotmail.com To: freeswitch-users at lists.freeswitch.org Subject: RE: Transmit fax locally for test Date: Mon, 22 Jun 2009 18:37:47 -0400 8000 is a local extension defined in the default dialplan. Tim > ------------------------------ > > Message: 2 > Date: Mon, 22 Jun 2009 15:05:20 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > what is 8000? is it local or is it a remote endpoint? > > /b > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > test.tif) > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > Insert movie times and more without leaving Hotmail?. See how. _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing? now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try_bing_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/e0d0be43/attachment.html From Richard.Lamkin at mettoni.com Tue Jun 23 15:45:47 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 23 Jun 2009 23:45:47 +0100 Subject: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7503@nickel.mettonigroup.com> Brian, Thank you for suggesting I try PARK. I tried PARK but unfortunately it sends out a 183 with SDP which stops the originator hearing ringing (ring back). If you know of a way to park without sending a 183 that would solve my problem. Regards Richard Lamkin Richard.lamkin at mettoni.com From: Brian West [mailto:brian at freeswitch.org] Sent: 23 June 2009 17:37 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem with handling unanswered calls for amanaged redirect On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote: Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. See below.. ie park. You should get an event via event socket you can decide what to do. 2 - I do not want that incoming call to be answered but just stay ringing. Can't really do it that way.. you can answer it but then you're responsible for generating ringback. And billing starts when you answer it. 3 - Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. Try using park ... this way you put the call in limbo and you can send the call commands at your leisure. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park So far I've managed 1 - I see on the incoming call on the event API 2 - I used sleep 180000 (3 mins) see rule below. 3 - failed - because the rule is executing a sleep command and I cannot break in with my redirect. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/9c1f38f0/attachment.html From msc at freeswitch.org Tue Jun 23 16:14:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 16:14:11 -0700 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: <87f2f3b90906231614t6223f65cr64e3dc492564a83c@mail.gmail.com> Is 8000 just a dialplan extension? I'm curious about the whole 8000 at 192.168.10.35 thing. I doubt that's necessary. For kicks try something like this: originate loopback/8000 &txfax(storage/fax/test.tif) That will drop the A leg right into extension 8000. -MC On Tue, Jun 23, 2009 at 3:12 PM, Tim B wrote: > Did anyone have any suggestions on this? Just to reiterate... > > - 8000 is a local extension defined in the default dialplan... see > http://pastebin.freeswitch.org/9450 for definition > > - didn't work: originate sofia/default/8000 at 192.168.10.35&txfax(storage/fax/test.tif) ... see > http://pastebin.freeswitch.org/9440 for log > > - had to add the FS ip (192.168.10.35) to the domains acl... now it to > works > > > > > > > Is this the proper way to configure? > > > Tim > > ------------------------------ > From: timb0311 at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: RE: Transmit fax locally for test > Date: Mon, 22 Jun 2009 18:37:47 -0400 > > 8000 is a local extension defined in the default dialplan. > > Tim > > > > ------------------------------ > > > > Message: 2 > > Date: Mon, 22 Jun 2009 15:05:20 -0400 > > From: Brian West > > Subject: Re: [Freeswitch-users] Transmit fax locally for test > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > > Content-Type: text/plain; charset="us-ascii" > > > > what is 8000? is it local or is it a remote endpoint? > > > > /b > > > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > > test.tif) > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------ > Insert movie times and more without leaving Hotmail?. See how. > ------------------------------ > Microsoft brings you a new way to search the web. Try Bing? now > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6df939cc/attachment-0001.html From otrcomm at isp-systems.net Tue Jun 23 17:57:49 2009 From: otrcomm at isp-systems.net (murrah boswell) Date: Tue, 23 Jun 2009 17:57:49 -0700 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <87f2f3b90906231255iecb6d4fm4635290051f9cf23@mail.gmail.com> References: <4A404CAA.5080809@isp-systems.net> <87f2f3b90906231255iecb6d4fm4635290051f9cf23@mail.gmail.com> Message-ID: <4A417A0D.4000209@isp-systems.net> > Curious - what kinds of SIP phones do the clients support? Have you decided > what you'd be using? > -MC I am still experimenting! I have a zoiper 2.0 installed on one of my test clients. zoiper seems to work fine, so now I am attempting to get the freeswitch/zoiper interface working. I will also try to get an ekiga working, but first the zoiper. First I have to figure out how to get freeswitch operational and will be working on that tonight! Regards, Murrah Boswell > > On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell wrote: > >> Hello All, >> >> I am an absolute newbee in the voip world but have a project where I >> believe freeswitch will work and need very, very basic guidance >> on how to setup a testbed in a thin client environment. >> >> I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and >> utilize fl_teachertool 0.07 to monitor the connected >> terminal clients (TCs). If you are not familiar with fl_teachertool, it >> allows a teacher to view thumbnail images of each TC logged >> in to the server. The teacher can click on any thumbnail and enlarge the >> view, monitor all applications running on a given TC, and >> take control of the keyboard and mouse of the TC. These are just a few of >> the capabilities of fl_teachertool. >> >> What I want to do is allow the teacher to establish voice communication >> using headsets and microphones with any one of the TCs by >> making a "phone" call via ethernet based upon ip of the TC through >> freeswitch using a softphone. >> >> Does this sound like something that is possible using freeswitch? If so, >> could someone please give me very basic instructions on how >> to setup this proof of concept? If I can just get a "teacher" stationed at >> my server talking to one "student" at a TC, I believe I >> can go from there. Currently I have a voiper softphone that functions, I >> believe, under gnome, but I have no idea how to configure >> the voiper to initiate calls through freeswitch or how to configure >> freeswitch to route the call to one of my TCs. >> >> I also need to keep this system fully self contained. That is, I can not >> have a requirement to use an outside sip service provider. >> >> Also, I would use any other linux sip softphones known to work with >> freeswitch that people feel would work better than a voiper. >> voiper seems to be more windows and mac based. I would really like to use >> an ekiga since they seem to be more linux based, but I do >> not believe that they have been thoroughly tested with freeswitch. >> >> Any help would be greatly appreciated! >> >> >> Regards, >> Murrah Boswell >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From timb0311 at hotmail.com Tue Jun 23 18:23:25 2009 From: timb0311 at hotmail.com (Tim B) Date: Tue, 23 Jun 2009 21:23:25 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: Yeah 8000 is just a dialplan extension. That worked... originate loopback/8000 &txfax(storage/fax/test.tif) Thanks MC. I guess the loopback bypasses all the security stuff and jumps right into the dialplan looking for a matching # condition? Tim > ------------------------------ > > Message: 3 > Date: Tue, 23 Jun 2009 16:14:11 -0700 > From: Michael Collins > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906231614t6223f65cr64e3dc492564a83c at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Is 8000 just a dialplan extension? I'm curious about the whole > 8000 at 192.168.10.35 thing. I doubt that's necessary. For kicks try something > like this: > > originate loopback/8000 &txfax(storage/fax/test.tif) > > That will drop the A leg right into extension 8000. > > -MC > > On Tue, Jun 23, 2009 at 3:12 PM, Tim B wrote: > > > Did anyone have any suggestions on this? Just to reiterate... > > > > - 8000 is a local extension defined in the default dialplan... see > > http://pastebin.freeswitch.org/9450 for definition > > > > - didn't work: originate sofia/default/8000 at 192.168.10.35&txfax(storage/fax/test.tif) ... see > > http://pastebin.freeswitch.org/9440 for log > > > > - had to add the FS ip (192.168.10.35) to the domains acl... now it to > > works > > > > > > > > > > > > > > Is this the proper way to configure? > > > > > > Tim > > > > ------------------------------ > > From: timb0311 at hotmail.com > > To: freeswitch-users at lists.freeswitch.org > > Subject: RE: Transmit fax locally for test > > Date: Mon, 22 Jun 2009 18:37:47 -0400 > > > > 8000 is a local extension defined in the default dialplan. > > > > Tim > > > > > > > ------------------------------ > > > > > > Message: 2 > > > Date: Mon, 22 Jun 2009 15:05:20 -0400 > > > From: Brian West > > > Subject: Re: [Freeswitch-users] Transmit fax locally for test > > > To: freeswitch-users at lists.freeswitch.org > > > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > > > Content-Type: text/plain; charset="us-ascii" > > > > > > what is 8000? is it local or is it a remote endpoint? > > > > > > /b > > > > > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > > > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > > > test.tif) > > > > > > Brian West > > > brian at freeswitch.org > > > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > ------------------------------ > > Insert movie times and more without leaving Hotmail?. See how. > > ------------------------------ > > Microsoft brings you a new way to search the web. Try Bing? now > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6df939cc/attachment.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 231 > ************************************************* _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing? now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try_bing_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/cc6ad227/attachment.html From darklion11 at yahoo.com Tue Jun 23 19:10:20 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 23 Jun 2009 19:10:20 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> References: <24109532.post@talk.nabble.com> <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> Message-ID: <24177512.post@talk.nabble.com> Where can i find this logs? Michael Jerris wrote: > > Try turning up your logging level to debug to see why the call is > hanging up. > > Mike > > On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: > >> >> My freeswitch has a mysql database consists of freeswitch tables, >> registrations and nibblebill on mysql configured it correctly and >> working... >> Issue is when I call external ip's sometimes it works sometimes not? >> >> 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 >> switch_core_session_enable_heartbeat() sofia/internal/ >> 1006 at 116.5.231.40 >> setting session heartbeat to 1 second(s). >> 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 >> switch_core_standard_on_execute() Hangup >> sofia/internal/1006 at 116.50.231.72 >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 >> switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40 >> ) >> Ended >> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 >> switch_core_session_thread() Close Channel >> sofia/internal/1006 at 116.5.231.40 >> [CS_DESTROY] >> >> On my acl.conf.xml I allow ip 116.5.231.40 >> >> >> >> >> >> >> >> I put this on my external and internal profile >> >> param name="apply-inbound-acl" value="globals"/> >> >> And put auth-calls to false... >> >> Please help me am really near to my success here in freeswitch... >> Thanks... > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24177512.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From vince.freeswitch at hightek.org Tue Jun 23 19:15:30 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Tue, 23 Jun 2009 21:15:30 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> Message-ID: <20090624021530.GA749@quark.hightek.org> Thanks for the response Anthony. On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote: > You are way off base in a few places, let me see if I can clarify a bit. > > Here are at least 2 pointers: > > 1) The release tarballs do not come with bootstrap because they already are > bootstrapped. > 2) FreeSWITCH does not depend on system libs so all the stuff about apr is > barking up the wrong tree. > we build our own apr and apr-utils Interesting. I do not know why I got the errors I mentioned before then until I installed the exact versions of those packages it seemed to need. > I suggest you try latest svn trunk of FS and follow the BSD build guidelines > on the WIKI since you say > it's closely compatible. Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: ==================================================== Checked out the current trunk with svn. Patched /usr/include/sys/resource.h Since Dragonfly has fixed or will be fixing this future releases I patched the system header to add RLIMIT_AS rather than patching freeswitch to use RLIMIT_VMEM. Compilation still failed but there are significant improvements. bootstrap.sh seems to have been successful this time. I seems to have worked with the bsd shell this time. I also did not have to link make to gmake. It appears to have properly called gmake when building in sub-directories when gmake was run from the top. Configure completed successfully but there were these warnings: checking dlfcn.h usability... no checking dlfcn.h presence... yes configure: WARNING: dlfcn.h: present but cannot be compiled configure: WARNING: dlfcn.h: check for missing prerequisite headers? configure: WARNING: dlfcn.h: see the Autoconf documentation configure: WARNING: dlfcn.h: section "Present But Cannot Be Compiled" configure: WARNING: dlfcn.h: proceeding with the preprocessor's result configure: WARNING: dlfcn.h: in the future, the compiler will take precedence checking for dlfcn.h... yes I do not know if this is going to cause a problem. I did not have to use the "--build=i386" option to configure this time. Compiling ========= Still lots of warnings of: warning: return makes pointer from integer without a cast Errors: It is apparently not checking return codes from make. It continues even when there are errors. Is this intentional?? su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features LTCOMPILE features.lo ... Making all in sresolv LTCOMPILE sres.lo LTCOMPILE sres_cache.lo LTCOMPILE sres_blocking.lo LTCOMPILE sresolv.lo LTCOMPILE sres_sip.lo sres_sip.c: In function `sres_sip_new': sres_sip.c:267: warning: return makes pointer from integer without a cast gmake[8]: *** [sres_sip.lo] Error 1 Making all in ipt LTCOMPILE base64.lo LTCOMPILE token64.lo LINK libipt.la ... There are about 12 errors of this nature before ending with Making all in nua LTCOMPILE nua.lo nua.c: In function `nua_create': nua.c:141: warning: return makes pointer from integer without a cast nua.c:144: warning: return makes pointer from integer without a cast gmake[9]: *** [nua.lo] Error 1 gmake[8]: *** [all] Error 2 gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. gmake[7]: *** [all-recursive] Error 1 Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 It says it has been successfully built. Apparently part of the same problem of not checking the return codes. It does not say what most of the errors are except for near the last when it says No rule to make target `iptsec/libiptsec.la' It just says "Error 1" or Error 2" which does not tell me what the problem is. From mcampbellsmith at gmail.com Tue Jun 23 19:41:54 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 24 Jun 2009 12:41:54 +1000 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Message-ID: <33c87fa30906231941m14d56dear6958288a54057137@mail.gmail.com> Thanks Anthony. I am getting closer. I had to put in the 146 address, which is the firewalled address I get at work. The problem now is that when the call is bridged, I do not hear audio. 2 scenarios: 1 -> the local extension is not registered. There is two way audio - I hear the voicemail in Gtalk and I can leave a message which can then be played back. 2 -> the local extension is registered. There is no audio In my incoming dialplan I am doing this bridge: It bridges okay, the phone rings, but there is no audio. On a side note: Isn't putting the candidate-acl list a temporary measure? When I travel, I will most likely get a different internal company IP address that does not start with 146. Isn't there a smarter way for dingaling to know that there is no RTP packets being received and then modify which candidate should be used? Thanks! > On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try adding this to your jingle in client.xml >> >> >> >> then edit acl.conf.xml and add this list >> >> >> ? >> ? >> ? >> >> >> this tells mod_dingaling that it should only pick candidates that pass the >> acl list given >> the one we made called wan excludes all the private ranges. >> >> If you update to latest trunk this list is created internally as "wan.auto" >> so you can use that >> instead of making one in your config. >> >> >> >> On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith < >> mcampbellsmith at gmail.com> wrote: >> >>> Hi! >>> >>> I am trying to call from my corporate network (firewalled) using Gtalk >>> to Freeswitch. ?I am not getting any audio. >>> >>> In the logs I see that mod_dingaling is using my internal corporate IP >>> address which is not publically addressable. >>> >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates >>> 146.xx.xx.xx:50320 >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable >>> Candidate 146.xx.xx.xx:50320 >>> >>> Further on in the log, I can see GTalk sending a new candidate IP >>> address to use: >>> 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 >>> name=rtp >>> type=local >>> protocol=udp >>> username=e+JTkVHT1xEkqXGD >>> password=fAxU6Pr1oF9Zq48U >>> address=192.168.1.102 >>> port=50322 >>> pref=1.00 >>> >>> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked >>> an IP [146.xx.xx.xx] >>> >>> and >>> >>> 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 >>> name=rtp >>> type=stun >>> protocol=udp >>> username=RBqyF2XNMYLfJNoU >>> password=DQMjon1fSVoJIRTp >>> address=124.xxx.xxx.xxx >>> port=50323 >>> pref=0.90 >>> >>> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked >>> an IP [146.xx.xx.xx] >>> and >>> >>> 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 >>> name=rtp >>> type=relay >>> protocol=udp >>> username=62L5zs2FHbcUdeCJ >>> password=KxmNgkUmZsLfuX6S >>> address=209.xx.xxx.xxx >>> port=19295 >>> pref=0.50 >>> >>> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked >>> an IP [146.xx.xx.xx] >>> >>> Because of this, I never get audio. ?Any ideas how to fix this? >>> >>> Thanks! From andrew at hijacked.us Tue Jun 23 20:19:51 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 23 Jun 2009 23:19:51 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090624021530.GA749@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> Message-ID: <20090624031950.GD2623@hijacked.us> On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: > Ok. I did this. > > Compilation still failed but there are significant improvements since > the last time. > > Here is what I did and the results: > It looks like some the games that sofia plays with errno makes Dragonfly unhappy. I also noticed that where the code checks for BSD-like systems (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is omitted, so obviously one of the first steps would be to fix that (if applicable). If you disable mod_sofia in modules conf, do the rest of the default modules build OK? For the record, DragonFly and FreeBSD have rather seriously diverged at this point, DragonFly forked from FreeBSD back in the 4.10 days or so and has changed a *lot* of things since, so I don't think it's gonna be quite as easy as you expected (but it's far from impossible either). Andrew From mcampbellsmith at gmail.com Tue Jun 23 20:30:42 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 24 Jun 2009 13:30:42 +1000 Subject: [Freeswitch-users] email core dump Message-ID: <33c87fa30906232030w679a24a7r39516ac7fa74572b@mail.gmail.com> Thanks Brian, but still no luck with the email.. I have configured exim4 so that I can send messages from the command line using 'mail' command and these are sent successfully. I still get a core dump in the log when freeswitch is trying to send the mail: /bin/cat: write error: Broken pipe sh: line 1: 4492 Done(1) /bin/cat /tmp/mail.1245811149abdc 4493 Segmentation fault (core dumped) | /usr/local/bin/eximcompat.sh -t xxx at xx.com 2009-06-24 12:39:09.285351 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.1245811149abdc] to [xxx at xx.com] 2009-06-24 12:39:09.285351 [DEBUG] mod_voicemail.c:2491 Sending message to xxx at xx.com eximcompat.sh is as described on the wiki: freeswitch:/# cat /usr/local/bin/eximcompat.sh #!/bin/bash exec exim4 -t Any other thoughts? From: Brian West > Subject: Re: [Freeswitch-users] email core dump > To: freeswitch-users at lists.freeswitch.org > Message-ID: <7C7A8ED9-ECED-4100-87F6-0875C054EC64 at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > /b > > On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: > > > Hi! > > > > I am trying to email from > > 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore > > original codec. > > 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to > > 1000 at 192.168.0.20 > > /bin/cat: write error: Broken pipe > > sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. > > 124558382500b1 > > 11976 Segmentation fault (core dumped) | exim4 -t > myemail at xx.com > > 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file > > [/tmp/mail.12455810042c7f] to [myemail at xx.com] > > > > I can manually send an email to myself with exim4, but freeswitch > > fails. > > > > Any ideas what I have configured incorrectly? > > > > Thanks > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/b2d38148/attachment.html From mike at jerris.com Tue Jun 23 21:44:29 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Jun 2009 00:44:29 -0400 Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <24177512.post@talk.nabble.com> References: <24109532.post@talk.nabble.com> <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> <24177512.post@talk.nabble.com> Message-ID: <5ECAE2C0-82F0-4D2B-B2F3-AB86213D02A7@jerris.com> Please see the debugging pages on the wiki On Jun 23, 2009, at 10:10 PM, Edmar Cruz wrote: > > Where can i find this logs? > > Michael Jerris wrote: >> >> Try turning up your logging level to debug to see why the call is >> hanging up. >> >> Mike >> >> On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: >> >>> >>> My freeswitch has a mysql database consists of freeswitch tables, >>> registrations and nibblebill on mysql configured it correctly and >>> working... >>> Issue is when I call external ip's sometimes it works sometimes not? >>> >>> 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 >>> switch_core_session_enable_heartbeat() sofia/internal/ >>> 1006 at 116.5.231.40 >>> setting session heartbeat to 1 second(s). >>> 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 >>> switch_core_standard_on_execute() Hangup >>> sofia/internal/1006 at 116.50.231.72 >>> [CS_EXECUTE] [NORMAL_CLEARING] >>> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 >>> switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40 >>> ) >>> Ended >>> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1006 at 116.5.231.40 >>> [CS_DESTROY] >>> >>> On my acl.conf.xml I allow ip 116.5.231.40 >>> >>> >>> >>> >>> >>> >>> >>> I put this on my external and internal profile >>> >>> param name="apply-inbound-acl" value="globals"/> >>> >>> And put auth-calls to false... >>> >>> Please help me am really near to my success here in freeswitch... >>> Thanks... >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24177512.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 23 21:53:23 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Jun 2009 00:53:23 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090624021530.GA749@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> Message-ID: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> On Jun 23, 2009, at 10:15 PM, Vincent Stemen wrote: > Thanks for the response Anthony. > > On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote: >> You are way off base in a few places, let me see if I can clarify a >> bit. >> >> Here are at least 2 pointers: >> >> 1) The release tarballs do not come with bootstrap because they >> already are >> bootstrapped. >> 2) FreeSWITCH does not depend on system libs so all the stuff about >> apr is >> barking up the wrong tree. >> we build our own apr and apr-utils > > Interesting. I do not know why I got the errors I mentioned before > then > until I installed the exact versions of those packages it seemed to > need. > > >> I suggest you try latest svn trunk of FS and follow the BSD build >> guidelines >> on the WIKI since you say >> it's closely compatible. > > Ok. I did this. > > Compilation still failed but there are significant improvements since > the last time. > > Here is what I did and the results: > > ==================================================== > Checked out the current trunk with svn. > > Patched /usr/include/sys/resource.h > > Since Dragonfly has fixed or will be fixing this future releases I > patched the > system header to add RLIMIT_AS rather than patching freeswitch to use > RLIMIT_VMEM. Can we make a patch ifdefing on RLIMIT_AS to make this always work without patches to system header files? > > > Compilation still failed but there are significant improvements. > > bootstrap.sh seems to have been successful this time. > > I seems to have worked with the bsd shell this time. > I also did not have to link make to gmake. It appears to have > properly > called gmake when building in sub-directories when gmake was run > from the top. > > Configure completed successfully but there were these warnings: > > checking dlfcn.h usability... no > checking dlfcn.h presence... yes > configure: WARNING: dlfcn.h: present but cannot be compiled > configure: WARNING: dlfcn.h: check for missing prerequisite > headers? > configure: WARNING: dlfcn.h: see the Autoconf documentation > configure: WARNING: dlfcn.h: section "Present But Cannot Be > Compiled" > configure: WARNING: dlfcn.h: proceeding with the preprocessor's > result > configure: WARNING: dlfcn.h: in the future, the compiler will take > precedence > checking for dlfcn.h... yes > This is probably fine, it means what it says, it won't try to compile with them bit the issue should probably be reported to distro maintainers > I do not know if this is going to cause a problem. > > I did not have to use the "--build=i386" option to configure this > time. > > > Compiling > ========= > > Still lots of warnings of: > warning: return makes pointer from integer without a cast > > Errors: > It is apparently not checking return codes from make. It continues > even when > there are errors. Is this intentional?? > > su_alloc.c: In function `su_salloc': > su_alloc.c:1518: warning: return makes pointer from integer without > a cast > gmake[9]: *** [su_alloc.lo] Error 1 > gmake[8]: *** [all] Error 2 > Making all in features > LTCOMPILE features.lo > ... > > Making all in sresolv > LTCOMPILE sres.lo > LTCOMPILE sres_cache.lo > LTCOMPILE sres_blocking.lo > LTCOMPILE sresolv.lo > LTCOMPILE sres_sip.lo > sres_sip.c: In function `sres_sip_new': > sres_sip.c:267: warning: return makes pointer from integer without > a cast > gmake[8]: *** [sres_sip.lo] Error 1 > Making all in ipt > LTCOMPILE base64.lo > LTCOMPILE token64.lo > LINK libipt.la > ... > > There are about 12 errors of this nature before ending with > > Making all in nua > LTCOMPILE nua.lo > nua.c: In function `nua_create': > nua.c:141: warning: return makes pointer from integer without a cast > nua.c:144: warning: return makes pointer from integer without a cast > gmake[9]: *** [nua.lo] Error 1 > gmake[8]: *** [all] Error 2 > gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed > by `libsofia-sip-ua.la'. Stop. > gmake[7]: *** [all-recursive] Error 1 > Making all in packages > gmake[6]: *** [all-recursive] Error 1 > gmake[5]: *** [all] Error 2 > gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- > ua.la] Error 2 > gmake[3]: *** [mod_sofia-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + gmake install + > +----------------------------------------------+ > gmake[1]: *** [all-recursive] Error 1 > gmake: *** [all] Error 2 > Can you post a bug to Jira.freeswitch.org with all these warnings, even better with patches to fix it. > > It says it has been successfully built. Apparently part of the same > problem of > not checking the return codes. > Patches to fix this appreciated > It does not say what most of the errors are except for near the last > when it > says > No rule to make target `iptsec/libiptsec.la' > > It just says "Error 1" or Error 2" which does not tell me what the > problem is. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dome at tel.co.th Tue Jun 23 22:36:01 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 24 Jun 2009 12:36:01 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway Message-ID: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> Dear All, Look like nibblebill does't work with multiple gatreway. I try > > > > Is 8000 just a dialplan extension? I'm curious about the whole > > 8000 at 192.168.10.35 thing. I doubt that's necessary. For kicks try > something > > like this: > > > > originate loopback/8000 &txfax(storage/fax/test.tif) > > > > That will drop the A leg right into extension 8000. > > > > -MC > > > > On Tue, Jun 23, 2009 at 3:12 PM, Tim B wrote: > > > > > Did anyone have any suggestions on this? Just to reiterate... > > > > > > - 8000 is a local extension defined in the default dialplan... see > > > http://pastebin.freeswitch.org/9450 for definition > > > > > > - didn't work: originate sofia/default/8000 at 192.168.10.35&txfax(storage/fax/test.tif) > ... see > > > http://pastebin.freeswitch.org/9440 for log > > > > > > - had to add the FS ip (192.168.10.35) to the domains acl... now it to > > > works > > > > > > > > > > > > > > > > > > > > > Is this the proper way to configure? > > > > > > > > > Tim > > > > > > ------------------------------ > > > From: timb0311 at hotmail.com > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: RE: Transmit fax locally for test > > > Date: Mon, 22 Jun 2009 18:37:47 -0400 > > > > > > 8000 is a local extension defined in the default dialplan. > > > > > > Tim > > > > > > > > > > ------------------------------ > > > > > > > > Message: 2 > > > > Date: Mon, 22 Jun 2009 15:05:20 -0400 > > > > From: Brian West > > > > Subject: Re: [Freeswitch-users] Transmit fax locally for test > > > > To: freeswitch-users at lists.freeswitch.org > > > > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > > > > Content-Type: text/plain; charset="us-ascii" > > > > > > > > what is 8000? is it local or is it a remote endpoint? > > > > > > > > /b > > > > > > > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > > > > > > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > > > > test.tif) > > > > > > > > Brian West > > > > brian at freeswitch.org > > > > > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > > > > > > > ------------------------------ > > > Insert movie times and more without leaving Hotmail?. See how.< > http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 > > > > > ------------------------------ > > > Microsoft brings you a new way to search the web. Try Bing? now< > http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try+bing_1x1 > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6df939cc/attachment.html > > > > ------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > End of Freeswitch-users Digest, Vol 36, Issue 231 > > ************************************************* > > > ------------------------------ > Microsoft brings you a new way to search the web. Try Bing? now > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/5217ad79/attachment.html From dome at tel.co.th Tue Jun 23 23:47:50 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 24 Jun 2009 13:47:50 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> Message-ID: <8ccbff060906232347i62290b26lacad4201431759f2@mail.gmail.com> 2009/6/24 Darren Schreiber > Shouldn't you be using {} not [] ? > you mean {nibble_rate=0.3}sofia/external/xxx at xxx.xxx.xxx.xxx |{nibble_rate=0.5}sofia/external/xxx at xxx.xxx.xxx.xxx I think {} use for all channel but [] for per channel Dome C. > ------------------------------ > *From:* Dome Charoenyost [mailto:dome at tel.co.th] > *Sent:* Tuesday, June 23, 2009 10:36 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Nibblebill and multiple gateway > > Dear All, > > Look like nibblebill does't work with multiple gatreway. > I try > data="nibble_account=0838833133"/> > > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx > |[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> > > nibblebill not found nibble_rate > > But > > data="nibble_account=0838833133"/> > > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx > |sofia/external/6626734000 at 202.xxx.xxx.xxx> > > Work fine > > What's difference from set application and [] ? > > Best Regards. > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/a972d7ba/attachment.html From mcampbellsmith at gmail.com Wed Jun 24 00:26:47 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 24 Jun 2009 17:26:47 +1000 Subject: [Freeswitch-users] freeswitch segfault Message-ID: <33c87fa30906240026x7dc92f3aida6b963ee8b1cd81@mail.gmail.com> Hi! My call dropped and I saw this error in the syslog: Jun 24 17:05:04 freeswitch kernel: [157531.309017] freeswitch[4621]: segfault at c ip b73b2a42 sp b72a3840 error 4 in mod_sofia.so[b7369000+16c000] How can I get more information on this fault to file a bug report? Thanks! From jason at jasonjgw.net Wed Jun 24 00:47:50 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 17:47:50 +1000 Subject: [Freeswitch-users] freeswitch segfault In-Reply-To: <33c87fa30906240026x7dc92f3aida6b963ee8b1cd81@mail.gmail.com> References: <33c87fa30906240026x7dc92f3aida6b963ee8b1cd81@mail.gmail.com> Message-ID: <20090624074750.GA11226@jdc.jasonjgw.net> Mark Campbell-Smith wrote: > How can I get more information on this fault to file a bug report? See the debugging FreeSWITCH page on the wiki, and set in the FreeSWITCH core configuration (by default in switch.conf.xml), or use a ulimit -c unlimited command before running FreeSWITCH. Next time it happens, apply gdb to the core file to obtain backtraces as described on the wiki. From darklion11 at yahoo.com Wed Jun 24 01:35:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 01:35:41 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Documentation Message-ID: <24180754.post@talk.nabble.com> HI, Is there any available complete documentation for Freeswitch with matching samples aside from wiki that works. With working samples like dialplans, outbounds and prefer codecs etc. Thanks... -- View this message in context: http://www.nabble.com/Freeswitch-Documentation-tp24180754p24180754.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Wed Jun 24 01:48:53 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 18:48:53 +1000 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <24180754.post@talk.nabble.com> References: <24180754.post@talk.nabble.com> Message-ID: <20090624084853.GA18344@jdc.jasonjgw.net> Edmar Cruz wrote: > > Is there any available complete documentation for Freeswitch with matching > samples aside from wiki that works. With working samples like dialplans, > outbounds and prefer codecs etc. There isn't much besides the wiki. Most of the documentation effort has been devoted to improving the wiki rather than writing external documentation. If you would like to contribute to the wiki, you are welcome to help the community by further improving and expanding the FreeSWITCH documentation available there. From darklion11 at yahoo.com Wed Jun 24 02:10:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 02:10:13 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? Message-ID: <24181208.post@talk.nabble.com> when a type on the API of freeswitch originate sofia/external/8011104 at 116.541.23.12 1001, 8011104 my extension that I want to call and its ip is 116.541.23.12. I register on 1001 using a softphone (X-Lite) and my ip is 116.541.23.11. It works actually. but when dialing on softphone 1001 account on ip 116.541.23.11 Temporary Unavailable... What do you think is the possible issue? On originate it works sometimes but an ERR - SERVICE_NOT_IMPLEMENTED Here is my dialplan on sip_profiles/external/myprofile.xml Set acl perfectly. I set auth-calls to false. No found error on logs. Just destroying the call... Please help me... Thanks -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181208.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Wed Jun 24 02:40:42 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Wed, 24 Jun 2009 17:40:42 +0800 Subject: [Freeswitch-users] mod_dingaling no audio Message-ID: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> Hi Guys, Here's my situation: The freeswitch server and my machine are behind the same LAN. If I commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by *originate dingaling/gmail.com/userAAA at gmail.com &echo*). However, external calls have no sound at all no matter whether this param is commented out or not. Please kindly let me know what other params to set to resolve this issue. Thanks, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/afc51acb/attachment.html From jason at jasonjgw.net Wed Jun 24 02:45:17 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 19:45:17 +1000 Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24181208.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> Message-ID: <20090624094517.GA22279@jdc.jasonjgw.net> Edmar Cruz wrote: > Here is my dialplan on sip_profiles/external/myprofile.xml > > > > > References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> Message-ID: search wiki from sth. like disable_rtp_autoajust , I don't remember the exact var. On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: > Hi Guys, > > Here's my situation: > > The freeswitch server and my machine are behind the same LAN. If I > commented out "ext-rtp-ip" from client.xml, I'm able to hear the > echo (by originate dingaling/gmail.com/userAAA at gmail.com &echo). > > However, external calls have no sound at all no matter whether this > param is commented out or not. > > Please kindly let me know what other params to set to resolve this > issue. > > Thanks, > -Jingwei > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/8a144b5c/attachment.html From darklion11 at yahoo.com Wed Jun 24 03:02:21 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 03:02:21 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <20090624094517.GA22279@jdc.jasonjgw.net> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> Message-ID: <24181933.post@talk.nabble.com> Ooops.. Sorry wrong spelling... Same issue Jason White-14 wrote: > > Edmar Cruz wrote: > >> Here is my dialplan on sip_profiles/external/myprofile.xml >> >> >> >> >> > The above should be $1 not @1 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Claudio.Cavalera at italtel.it Wed Jun 24 03:35:32 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 24 Jun 2009 12:35:32 +0200 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> Message-ID: > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > play with it from the cli > freeswitch> eval ${foo:-4:4} > API CALL [eval(${foo:-4:4})] output: > 2345 Thanks anthm! This way to test directly on the cli is really nice, I didn't know of it. If I've understood well, a negative number as first parameter will make the second parameter useless/meaningless. Besides, the second parameter is useless/meaningless if set equal to 0 or lesser than 0. I still think that we could benefit from more power here in the dialplan, if I have 12346578 at domain.org I'm not able to grab out the number (of which I don't know the length) even if I know that domain.org is always the same length. I've looked for something useful in http://apr.apache.org/docs/apr/1.3/apr__strings_8h.html and switch_apr.c but I think I'll end up calling an external script. :) Best Regars, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From saeedahmad1981 at gmail.com Wed Jun 24 03:53:30 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 24 Jun 2009 12:53:30 +0200 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: References: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> Message-ID: <2200BAECC6FD423784C068B2E0595BA7@saeedlaptop> Can we also test dialplan using CLI, like "dial" in asterisk? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cavalera Claudio Luigi Sent: Wednesday, June 24, 2009 12:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Variable manipulation in the dialplan > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > play with it from the cli > freeswitch> eval ${foo:-4:4} > API CALL [eval(${foo:-4:4})] output: > 2345 Thanks anthm! This way to test directly on the cli is really nice, I didn't know of it. If I've understood well, a negative number as first parameter will make the second parameter useless/meaningless. Besides, the second parameter is useless/meaningless if set equal to 0 or lesser than 0. I still think that we could benefit from more power here in the dialplan, if I have 12346578 at domain.org I'm not able to grab out the number (of which I don't know the length) even if I know that domain.org is always the same length. I've looked for something useful in http://apr.apache.org/docs/apr/1.3/apr__strings_8h.html and switch_apr.c but I think I'll end up calling an external script. :) Best Regars, Claudio Internet Email Confidentiality Footer ---------------------------------------------------------------------------- ------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ---------------------------------------------------------------------------- ------------------------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jason at jasonjgw.net Wed Jun 24 04:04:13 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 21:04:13 +1000 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: <2200BAECC6FD423784C068B2E0595BA7@saeedlaptop> References: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> <2200BAECC6FD423784C068B2E0595BA7@saeedlaptop> Message-ID: <20090624110413.GA27336@jdc.jasonjgw.net> Saeed Ahmed wrote: > Can we also test dialplan using CLI, like "dial" in asterisk? Have a look at the originate command. From dftoro at yahoo.com Wed Jun 24 06:07:16 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 24 Jun 2009 06:07:16 -0700 (PDT) Subject: [Freeswitch-users] transfer_ringback from mod_managed Message-ID: <413467.8366.qm@web33501.mail.mud.yahoo.com> Greetings ? When I use Session.SetVariable("transfer_ringback", "us-ring") from managed code the bridge fails with "NO_ANSWER" cause. If I use from xml dial plan the call is stablished. ? I have FS rev 13750 running on Windows. ? This is a issue or I don't use properly transfer_ringback variable ? ? Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/562d4c7b/attachment.html From brian at freeswitch.org Wed Jun 24 06:32:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 24 Jun 2009 08:32:00 -0500 Subject: [Freeswitch-users] transfer_ringback from mod_managed In-Reply-To: <413467.8366.qm@web33501.mail.mud.yahoo.com> References: <413467.8366.qm@web33501.mail.mud.yahoo.com> Message-ID: Well its the same you use ${us-ring} in both cases. /b On Jun 24, 2009, at 8:07 AM, Diego Toro wrote: > Greetings > > When I use Session.SetVariable("transfer_ringback", "us-ring") from > managed code the bridge fails with "NO_ANSWER" cause. If I use > from > xml dial plan the call is stablished. > > I have FS rev 13750 running on Windows. > > This is a issue or I don't use properly transfer_ringback variable ? > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/6406a6ff/attachment.html From dftoro at yahoo.com Wed Jun 24 07:20:52 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 24 Jun 2009 07:20:52 -0700 (PDT) Subject: [Freeswitch-users] transfer_ringback from mod_managed Message-ID: <936224.63343.qm@web33503.mail.mud.yahoo.com> Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring}); I have message: "[CRIT] switch_channel.c:633 Invalid data (${transfer_ringback} contains a variable)". ? Using from managed code: ? string stUsRing = _Session.GetVariable("us-ring"); Session.SetVariable("ringback", stUsRing); Session.SetVariable("transfer_ringback", stUsRing); ? The bridge works fine. ? The question is, using Session is not possible acces directly global vars way ${var_name} ? ? Thanks ? Diego --- On Wed, 6/24/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] transfer_ringback from mod_managed To: freeswitch-users at lists.freeswitch.org Date: Wednesday, June 24, 2009, 8:32 AM Well its the same you use ${us-ring} in both cases. /b On Jun 24, 2009, at 8:07 AM, Diego Toro wrote: Greetings ? When I use Session.SetVariable("transfer_ringback", "us-ring") from managed code the bridge fails with "NO_ANSWER" cause. If I use from xml dial plan the call is stablished. ? I have FS rev 13750 running on Windows. ? This is a issue or I don't use properly transfer_ringback variable ? ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/5c35cd9e/attachment.html From brian at freeswitch.org Wed Jun 24 07:33:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 24 Jun 2009 09:33:04 -0500 Subject: [Freeswitch-users] transfer_ringback from mod_managed In-Reply-To: <936224.63343.qm@web33503.mail.mud.yahoo.com> References: <936224.63343.qm@web33503.mail.mud.yahoo.com> Message-ID: <775DF765-03C2-4F49-BD23-E2E9BEB2085A@freeswitch.org> Chances are you need to get var us-ring then use that to set the transfer_ringback /b On Jun 24, 2009, at 9:20 AM, Diego Toro wrote: > Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring}); > I have message: "[CRIT] switch_channel.c:633 Invalid data ($ > {transfer_ringback} contains a variable)". > > Using from managed code: > > string stUsRing = _Session.GetVariable("us-ring"); > Session.SetVariable("ringback", stUsRing); > Session.SetVariable("transfer_ringback", stUsRing); > > The bridge works fine. > > The question is, using Session is not possible acces directly global > vars way ${var_name} ? > > Thanks > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/8a8fb06f/attachment.html From mike at jerris.com Wed Jun 24 08:15:01 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Jun 2009 11:15:01 -0400 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> Message-ID: <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> try adding before the bridge and report back results. Mike On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: > Dear All, > > Look like nibblebill does't work with multiple gatreway. > I try > > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx > |[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> > > nibblebill not found nibble_rate > > But > > > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx > |sofia/external/6626734000 at 202.xxx.xxx.xxx> > > Work fine > > What's difference from set application and [] ? > > Best Regards. > Dome C. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/561334f5/attachment-0001.html From Richard.Lamkin at mettoni.com Wed Jun 24 08:29:34 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Wed, 24 Jun 2009 16:29:34 +0100 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg call-command: execute execute-app-name: redirect execute-app-arg: sip:@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg call-command: execute execute-app-name: hangup execute-app-arg: Using command "api show channels" I find the following held on FS The only way I've found to remove these calls is "api hupall" ------------------------- uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. ------------------- The SIP signalling is correct with an outgoing "302 moved temporarily" [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either "api hupall", or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to "api hupall". "api hupall" kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lamkin at mettoni.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/47300618/attachment.html From msc at freeswitch.org Wed Jun 24 08:57:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Jun 2009 08:57:24 -0700 Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <24177512.post@talk.nabble.com> References: <24109532.post@talk.nabble.com> <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> <24177512.post@talk.nabble.com> Message-ID: <87f2f3b90906240857k25f38864r60b305fdbc24fdc1@mail.gmail.com> On Tue, Jun 23, 2009 at 7:10 PM, Edmar Cruz wrote: > > Where can i find this logs? > Please look at this page because it will give you a lot of information about how to collect information for debugging: http://wiki.freeswitch.org/wiki/Reporting_Bugs I recommend setting aside 20 minutes to read that page and learn how to turn on debugging, capture command line interface output, etc. If you master those basic skills it will save you (and us) a lot of time. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/01b98cbe/attachment.html From freeswitch-users at digitaldan.com Wed Jun 24 09:10:04 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 24 Jun 2009 10:10:04 -0600 (MDT) Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <4A417A0D.4000209@isp-systems.net> Message-ID: <9827933.1121245859861542.JavaMail.daniel@radio> I was thinking, for the students machines your could run the command line program linphonec in auto answer mode ( linphonec -a ). When a teacher calls one of the thin clients, the student would automatically hear and be able to speak to the teacher and not have to worry about dealing with a softphone. You could probably do that as well with freeswitch running as a client on each thin client, but linphoncc does this very well today. D- ----- Original Message ----- From: "murrah boswell" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 23, 2009 6:57:49 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Newbee: Need Help Thin Client Environment > Curious - what kinds of SIP phones do the clients support? Have you decided > what you'd be using? > -MC I am still experimenting! I have a zoiper 2.0 installed on one of my test clients. zoiper seems to work fine, so now I am attempting to get the freeswitch/zoiper interface working. I will also try to get an ekiga working, but first the zoiper. First I have to figure out how to get freeswitch operational and will be working on that tonight! Regards, Murrah Boswell > > On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell wrote: > >> Hello All, >> >> I am an absolute newbee in the voip world but have a project where I >> believe freeswitch will work and need very, very basic guidance >> on how to setup a testbed in a thin client environment. >> >> I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and >> utilize fl_teachertool 0.07 to monitor the connected >> terminal clients (TCs). If you are not familiar with fl_teachertool, it >> allows a teacher to view thumbnail images of each TC logged >> in to the server. The teacher can click on any thumbnail and enlarge the >> view, monitor all applications running on a given TC, and >> take control of the keyboard and mouse of the TC. These are just a few of >> the capabilities of fl_teachertool. >> >> What I want to do is allow the teacher to establish voice communication >> using headsets and microphones with any one of the TCs by >> making a "phone" call via ethernet based upon ip of the TC through >> freeswitch using a softphone. >> >> Does this sound like something that is possible using freeswitch? If so, >> could someone please give me very basic instructions on how >> to setup this proof of concept? If I can just get a "teacher" stationed at >> my server talking to one "student" at a TC, I believe I >> can go from there. Currently I have a voiper softphone that functions, I >> believe, under gnome, but I have no idea how to configure >> the voiper to initiate calls through freeswitch or how to configure >> freeswitch to route the call to one of my TCs. >> >> I also need to keep this system fully self contained. That is, I can not >> have a requirement to use an outside sip service provider. >> >> Also, I would use any other linux sip softphones known to work with >> freeswitch that people feel would work better than a voiper. >> voiper seems to be more windows and mac based. I would really like to use >> an ekiga since they seem to be more linux based, but I do >> not believe that they have been thoroughly tested with freeswitch. >> >> Any help would be greatly appreciated! >> >> >> Regards, >> Murrah Boswell >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/13ffbb0b/attachment-0001.html From Richard.Lamkin at mettoni.com Wed Jun 24 09:54:39 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Wed, 24 Jun 2009 17:54:39 +0100 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7887@nickel.mettonigroup.com> I have also observed that the cpu load goes up to 100% when only a couple of orphaned calls are left without being cleared by "api hupall". Richard Lamkin richard.lamkin at mettoni.com From: Richard Lamkin Sent: 24 June 2009 16:30 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg call-command: execute execute-app-name: redirect execute-app-arg: sip:@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg call-command: execute execute-app-name: hangup execute-app-arg: Using command "api show channels" I find the following held on FS The only way I've found to remove these calls is "api hupall" ------------------------- uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. ------------------- The SIP signalling is correct with an outgoing "302 moved temporarily" [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either "api hupall", or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to "api hupall". "api hupall" kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lamkin at mettoni.com ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/3fdf0f98/attachment.html From anthony.minessale at gmail.com Wed Jun 24 10:36:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 24 Jun 2009 12:36:31 -0500 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C7887@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF46138053C7887@nickel.mettonigroup.com> Message-ID: <191c3a030906241036m7c0ff8d7x6170f2e76d00db1c@mail.gmail.com> can do the following: 1) "make current" or do a fresh checkout to make sure you build is clean. 2) try executing the app "ring_ready" with no args in place of respond 180 and see if it makes any difference. 3) clear out your logfile by stopping FS and deleting /usr/local/freeswitch/log/freeswitch.log and reproduce. Then send me the log along with the list of channels you still see stuck. report the findings to jira http://jira.freeswitch.org and let me know the ticket number. make sure all your attachments have a .txt extensions when they are text files as jira has a bug of it's own with attachments and file types. On Wed, Jun 24, 2009 at 11:54 AM, Richard Lamkin wrote: > I have also observed that the cpu load goes up to 100% when only a couple > of orphaned calls are left without being cleared by ?api hupall?. > > > > Richard Lamkin > > > > richard.lamkin at mettoni.com > > > > > > *From:* Richard Lamkin > *Sent:* 24 June 2009 16:30 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Orphaned calls left on FS after redirect off > of FS > > > > I am using the API to manage calls as they arrive at FS from a trunk > > > > I have a very simple Dial plan rule that parks the incoming call. > > > > > > > > > > > > > > > > > > Once the call is parked via the API I first send a ringing (to keep the > originator happy) > > > > sendmsg > > call-command: execute > > execute-app-name: respond > > execute-app-arg: 180 > > > > Via the API I then redirect the call on to another PSTN number back through > the same gateway > > > > sendmsg > > call-command: execute > > execute-app-name: redirect > > execute-app-arg: sip:@194.0.147.16 > > > > The redirection works well and the originator and destination are connected > correctly. > > > > But after the call has left FS I?m still left with some call debris which I > cannot clear down using > > > > sendmsg > > call-command: execute > > execute-app-name: hangup > > execute-app-arg: > > > > > > Using command ?api show channels? I find the following held on FS The > only way I?ve found to remove these calls is ?api hupall? > > > > ------------------------- > > > uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate > > 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 > 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060 > ,CS_EXECUTE,0203196599,0203196599, > > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16 > ,XML,Public,PCMU,8000,PCMU,8000 > > c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 > 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060 > ,CS_EXECUTE,0203196598,0203196598, > > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16 > ,XML,Public,PCMU,8000,PCMU,8000 > > b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 > 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060 > ,CS_EXECUTE,0203196599,0203196599, > > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16 > ,XML,Public,PCMU,8000,PCMU,8000 > > 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 > 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060 > ,CS_EXECUTE,0189728400,0189728400, > > 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16 > ,XML,Public,PCMA,8000,PCMA,8000 > > > > 4 total. > > ------------------- > > > > The SIP signalling is correct with an outgoing ?302 moved temporarily? > [with the new destination in the contact] which is then Ack?ed by the > switch. From a SIP point of view the call no longer on FS. > > The only way I?ve found to remove these phantom calls is either ?api > hupall?, or restart the Sip profile. > > > > Any suggestions on how I can remove these phantom calls without recourse to > ?api hupall?. ?api hupall? kills any incoming calls as well as the stuck > calls. > > > > Regards > > > > Richard Lamkin > > richard.lamkin at mettoni.com > > > > > > > > > > > > > > > > > > ************************************************************************* > > Please consider the environment before printing this e-mail > > ************************************************************************* > > This email and any files transmitted with it are confidential and > > intended solely for the use of the individual or entity to whom they > > are addressed. If you have received this email in error please notify > > the system manager. http://www.mettoni.com > > > > Mettoni Ltd > > Registered in England and Wales: 4485956 > > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > > ************************************************************************* > > ************************************************************************* > Please consider the environment before printing this e-mail > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/ddb11d00/attachment-0001.html From msc at freeswitch.org Wed Jun 24 11:12:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Jun 2009 11:12:34 -0700 Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24181208.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> Message-ID: <87f2f3b90906241112h65775787ge540160db7cf62f4@mail.gmail.com> Turn on debugging and capture the output. Put it in a pastebin and post the link here. We'll take a look. -MC On Wed, Jun 24, 2009 at 2:10 AM, Edmar Cruz wrote: > > when a type on the API of freeswitch originate > sofia/external/8011104 at 116.541.23.12 1001, 8011104 my extension that I > want > to call and its ip is 116.541.23.12. I register on 1001 using a softphone > (X-Lite) and my ip is 116.541.23.11. It works actually. > > but when dialing on softphone 1001 account on ip 116.541.23.11 Temporary > Unavailable... > > What do you think is the possible issue? > > On originate it works sometimes but an ERR - SERVICE_NOT_IMPLEMENTED > > Here is my dialplan on sip_profiles/external/myprofile.xml > > > > > data="sofia/external/@1 at 116.541.23.12"/> > > > > Set acl perfectly. > > I set auth-calls to false. > > No found error on logs. > > Just destroying the call... > > Please help me... > > Thanks > > -- > View this message in context: > http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181208.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/f5101644/attachment.html From brian at freeswitch.org Wed Jun 24 11:20:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 24 Jun 2009 13:20:50 -0500 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> Message-ID: I have tried to reproduce this issue but haven't been able too... What SVN Rev are you on? /b On Jun 24, 2009, at 10:29 AM, Richard Lamkin wrote: > I am using the API to manage calls as they arrive at FS from a trunk > > I have a very simple Dial plan rule that parks the incoming call. > > > > > > > > > Once the call is parked via the API I first send a ringing (to keep > the originator happy) > > sendmsg > call-command: execute > execute-app-name: respond > execute-app-arg: 180 > > Via the API I then redirect the call on to another PSTN number back > through the same gateway > > sendmsg > call-command: execute > execute-app-name: redirect > execute-app-arg: sip:@194.0.147.16 > > The redirection works well and the originator and destination are > connected correctly. > > But after the call has left FS I?m still left with some call debris > which I cannot clear down using > > sendmsg > call-command: execute > execute-app-name: hangup > execute-app-arg: > > > Using command ?api show channels? I find the following held on FS > The only way I?ve found to remove these calls is ?api hupall? > > ------------------------- > uuid > ,created > ,created_epoch > ,name > ,state > ,cid_name > ,cid_num > ,ip_addr > ,dest > ,application > ,application_data > ,dialplan,context,read_codec,read_rate,write_codec,write_rate > 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16 > :5060,CS_EXECUTE,0203196599,0203196599, > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public,PCMU,8000,PCMU,8000 > c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16 > :5060,CS_EXECUTE,0203196598,0203196598, > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public,PCMU,8000,PCMU,8000 > b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16 > :5060,CS_EXECUTE,0203196599,0203196599, > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public,PCMU,8000,PCMU,8000 > 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16 > :5060,CS_EXECUTE,0189728400,0189728400, > 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public,PCMA,8000,PCMA,8000 > > 4 total. > ------------------- > > The SIP signalling is correct with an outgoing ?302 moved > temporarily? [with the new destination in the contact] which is then > Ack?ed by the switch. From a SIP point of view the call no longer > on FS. > The only way I?ve found to remove these phantom calls is either ?api > hupall?, or restart the Sip profile. > > Any suggestions on how I can remove these phantom calls without > recourse to ?api hupall?. ?api hupall? kills any incoming calls as > well as the stuck calls. > > Regards > > Richard Lamkin > richard.lamkin at mettoni.com > > > > > > > > > ************************************************************************* > Please consider the environment before printing this e-mail > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/fa9797f7/attachment-0001.html From john at feith.com Wed Jun 24 18:12:17 2009 From: john at feith.com (John Wehle) Date: Wed, 24 Jun 2009 21:12:17 -0400 (EDT) Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking Message-ID: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> I have a lua script that originates a call: local s = freeswitch.Session ( "{ignore_early_media=true,origination_caller_id_name=" .. caller .. "}loopback/" .. destination .. "/default/XML") s:execute ("sleep", "1000") ... which works fine if a valid number is supplied. However if a invalid number is supplied, then the script hits: [ERR] switch_cpp.cpp:607 session is not initalized [ERR] freeswitch_lua.cpp:102 session is not initalized What's the recommended way to check if the session constructor was successful (i.e. the number could be dialed)? My other option is: local s = freeswitch.Session () local r = s:originate (nil, "{ignore_early_media=true,origination_caller_id_name=" .. caller .. "}loopback/" .. destination .. "/default/XML", 300) if r == 1 then stream:write ("-ERR call failed\n") return end which does handle invalid numbers however there are the minor issues such as: a) The documentation seems to strongly discourage using the originate method for some reason. b) The lua originate method seems to require timeout to be specified even though the documentation implies it's optional. c) Using this approach causes the message: [WARNING] mod_limit.c:576 USAGE: hash [insert|delete]/// which I have yet to track down. Thoughts? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From jingwei.yang at gmail.com Wed Jun 24 19:23:19 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 10:23:19 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> Message-ID: <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put "disable-rtp-auto-adjust" inside client.xml. No matter what value this parameter has (true or false), local IP is able to hear the echo but external ones still have no audio. On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: > search wiki from sth. like disable_rtp_autoajust , I don't remember the > exact var. > > On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: > > Hi Guys, > > Here's my situation: > > The freeswitch server and my machine are behind the same LAN. If I > commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by > *originate dingaling/gmail.com/userAAA at gmail.com &echo*). > > However, external calls have no sound at all no matter whether this param > is commented out or not. > > Please kindly let me know what other params to set to resolve this issue. > > Thanks, > -Jingwei > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/96fff17b/attachment.html From darklion11 at yahoo.com Wed Jun 24 19:37:14 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 19:37:14 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <87f2f3b90906241112h65775787ge540160db7cf62f4@mail.gmail.com> References: <24181208.post@talk.nabble.com> <87f2f3b90906241112h65775787ge540160db7cf62f4@mail.gmail.com> Message-ID: <24196071.post@talk.nabble.com> This is on my freeswitch logs... 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 sofia_glue_tech_set_codec() Set Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_INIT 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State INIT 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at 116.541.23.12 SOFIA INIT 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> CS_ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State INIT going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State ROUTING 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 Standard ROUTING 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Edmar->639273642511 in context public Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->outside_call] continue=true Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition [outside_call] Dialplan: sofia/internal/1001 at 116.541.23.11 Action set(outside_call=true) Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->call_debug] continue=true Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->public_extensions] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_extensions] destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->public_did] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_did] destination_number(639273642511) =~ /^(5551212)$/ break=on-false 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1001 at 116.541.23.11) State Change CS_ROUTING -> CS_EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State ROUTING going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State EXECUTE 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 Standard EXECUTE EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1001 at 116.541.23.11 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/1001 at 116.541.23.11 [KILL] 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State EXECUTE going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_HANGUP 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State HANGUP 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 480 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State HANGUP going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State Change CS_HANGUP -> CS_REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11) State REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/1001 at 116.541.23.11 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11) State REPORTING going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State Change CS_REPORTING -> CS_DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11) Locked, Waiting on external entities 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11) Ended 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1001 at 116.541.23.11 [CS_DESTROY] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 116.541.23.11) State DESTROY 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 Standard DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 116.541.23.11) State DESTROY going to sleep -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24196071.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Wed Jun 24 19:52:50 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 10:52:50 +0800 Subject: [Freeswitch-users] FreeSwitch at backend Message-ID: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> Hi Folks, I understand freeSwitch is supporting a couple of languages for call controls like Lua, Javascript, Perl, Java... However, after digging into the detailed wiki pages, I found out the codes written in those languages can only be executed via the freeswitch console. I was wondering whether it's possible to run FreeSwitch at backend and have a piece of Java program (outside the freeSwitch console) to invoke freeSwitch commands? Thanks, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d71c574d/attachment.html From paul.degt at gmail.com Wed Jun 24 20:13:05 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Wed, 24 Jun 2009 23:13:05 -0400 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> Message-ID: <4A42EB41.8030703@gmail.com> You can use FS socket event interface for that. See free Java lib for inbound socket event here: http://versafon.com/versafonweb/Software.jsp Jingwei Yang wrote: > Hi Folks, > > I understand freeSwitch is supporting a couple of languages for call > controls like Lua, Javascript, Perl, Java... However, after digging > into the detailed wiki pages, I found out the codes written in those > languages can only be executed via the freeswitch console. I was > wondering whether it's possible to run FreeSwitch at backend and have > a piece of Java program (outside the freeSwitch console) to invoke > freeSwitch commands? > > Thanks, > -Jingwei > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Jun 24 20:24:55 2009 From: dujinfang at gmail.com (seven) Date: Thu, 25 Jun 2009 11:24:55 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> Message-ID: uncomment ext-rtp-ip On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: > Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and > put "disable-rtp-auto-adjust" inside client.xml. No matter what > value this parameter has (true or false), local IP is able to hear > the echo but external ones still have no audio. > > On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: > search wiki from sth. like disable_rtp_autoajust , I don't remember > the exact var. > > > On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >> Hi Guys, >> >> Here's my situation: >> >> The freeswitch server and my machine are behind the same LAN. If I >> commented out "ext-rtp-ip" from client.xml, I'm able to hear the >> echo (by originate dingaling/gmail.com/userAAA at gmail.com &echo). >> >> However, external calls have no sound at all no matter whether this >> param is commented out or not. >> >> Please kindly let me know what other params to set to resolve this >> issue. >> >> Thanks, >> -Jingwei >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a9cb6649/attachment-0001.html From dujinfang at gmail.com Wed Jun 24 20:26:10 2009 From: dujinfang at gmail.com (seven) Date: Thu, 25 Jun 2009 11:26:10 +0800 Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24181933.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> <24181933.post@talk.nabble.com> Message-ID: <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> put your extension in dialplan/public.xml instead of sip_profiles/ external/myprofile.xml On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: > > Ooops.. Sorry wrong spelling... Same issue > > Jason White-14 wrote: >> >> Edmar Cruz wrote: >> >>> Here is my dialplan on sip_profiles/external/myprofile.xml >>> >>> >>> >>> >>> > >> The above should be $1 not @1 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jingwei.yang at gmail.com Wed Jun 24 20:26:46 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 11:26:46 +0800 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <4A42EB41.8030703@gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> <4A42EB41.8030703@gmail.com> Message-ID: <13529f9d0906242026w21d8f0dcla822ec94bfb6a4e6@mail.gmail.com> Hi Paul, thanks for your reply. I've give it a try. On Thu, Jun 25, 2009 at 11:13 AM, paul.degt at gmail.com wrote: > You can use FS socket event interface for that. See free Java lib for > inbound socket event here: http://versafon.com/versafonweb/Software.jsp > > Jingwei Yang wrote: > > Hi Folks, > > > > I understand freeSwitch is supporting a couple of languages for call > > controls like Lua, Javascript, Perl, Java... However, after digging > > into the detailed wiki pages, I found out the codes written in those > > languages can only be executed via the freeswitch console. I was > > wondering whether it's possible to run FreeSwitch at backend and have > > a piece of Java program (outside the freeSwitch console) to invoke > > freeSwitch commands? > > > > Thanks, > > -Jingwei > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/4034fd1a/attachment.html From darklion11 at yahoo.com Wed Jun 24 20:41:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 20:41:01 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> <24181933.post@talk.nabble.com> <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> Message-ID: <24196467.post@talk.nabble.com> If then, what bridge i shall call to? Like this? dujinfang wrote: > > put your extension in dialplan/public.xml instead of sip_profiles/ > external/myprofile.xml > > > On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: > >> >> Ooops.. Sorry wrong spelling... Same issue >> >> Jason White-14 wrote: >>> >>> Edmar Cruz wrote: >>> >>>> Here is my dialplan on sip_profiles/external/myprofile.xml >>>> >>>> >>>> >>>> >>>> >> >>> The above should be $1 not @1 >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24196467.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Wed Jun 24 20:42:09 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 11:42:09 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> Message-ID: <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> Thanks seven. External IPs have sound echo this time with ext-rtp-ip uncommented and disable-rtp-auto-adjust=true. However, internal IP has no audio this time no matter what value disable-rtp-auto-adjust is... On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: > uncomment ext-rtp-ip > > On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: > > Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put > "disable-rtp-auto-adjust" inside client.xml. No matter what value this > parameter has (true or false), local IP is able to hear the echo but > external ones still have no audio. > > On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: > >> search wiki from sth. like disable_rtp_autoajust , I don't remember the >> exact var. >> >> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >> >> Hi Guys, >> >> Here's my situation: >> >> The freeswitch server and my machine are behind the same LAN. If I >> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >> >> However, external calls have no sound at all no matter whether this param >> is commented out or not. >> >> Please kindly let me know what other params to set to resolve this issue. >> >> Thanks, >> -Jingwei >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/3c45c8b1/attachment.html From chris.chen2004 at gmail.com Wed Jun 24 20:53:33 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 24 Jun 2009 23:53:33 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> Message-ID: <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> Please provide your client.xml detail with confidential information crossout, I have gtalk client and server working properly behind the NAT. I should be able to help you. Chris On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: > Thanks seven. External IPs have sound echo this time with ext-rtp-ip > uncommented and disable-rtp-auto-adjust=true. However, internal IP has no > audio this time no matter what value disable-rtp-auto-adjust is... > > > On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: > >> uncomment ext-rtp-ip >> >> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >> >> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put >> "disable-rtp-auto-adjust" inside client.xml. No matter what value this >> parameter has (true or false), local IP is able to hear the echo but >> external ones still have no audio. >> >> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >> >>> search wiki from sth. like disable_rtp_autoajust , I don't remember the >>> exact var. >>> >>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>> >>> Hi Guys, >>> >>> Here's my situation: >>> >>> The freeswitch server and my machine are behind the same LAN. If I >>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>> >>> However, external calls have no sound at all no matter whether this param >>> is commented out or not. >>> >>> Please kindly let me know what other params to set to resolve this issue. >>> >>> Thanks, >>> -Jingwei >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/95ffa9af/attachment-0001.html From jingwei.yang at gmail.com Wed Jun 24 22:31:35 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 13:31:35 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> Message-ID: <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> Hi Chris, thanks for your help. Here's my client.xml On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: > Please provide your client.xml detail with confidential information > crossout, I have gtalk client and server working properly behind the NAT. > I should be able to help you. > > Chris > > > On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: > >> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >> audio this time no matter what value disable-rtp-auto-adjust is... >> >> >> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >> >>> uncomment ext-rtp-ip >>> >>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>> >>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put >>> "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>> parameter has (true or false), local IP is able to hear the echo but >>> external ones still have no audio. >>> >>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>> >>>> search wiki from sth. like disable_rtp_autoajust , I don't remember the >>>> exact var. >>>> >>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>> >>>> Hi Guys, >>>> >>>> Here's my situation: >>>> >>>> The freeswitch server and my machine are behind the same LAN. If I >>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>> >>>> However, external calls have no sound at all no matter whether this >>>> param is commented out or not. >>>> >>>> Please kindly let me know what other params to set to resolve this >>>> issue. >>>> >>>> Thanks, >>>> -Jingwei >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a8de1d86/attachment.html From msc at freeswitch.org Wed Jun 24 22:38:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Jun 2009 22:38:21 -0700 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> Message-ID: <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> FYI, Any language that can establish a network socket and send/receive information over that socket can be used to control FS. FreeSWITCH comes with ESL - the event socket library - that can abstract away some of the grunt work, but there isn't a Java one that I'm aware of. -MC On Wed, Jun 24, 2009 at 7:52 PM, Jingwei Yang wrote: > Hi Folks, > > I understand freeSwitch is supporting a couple of languages for call > controls like Lua, Javascript, Perl, Java... However, after digging into the > detailed wiki pages, I found out the codes written in those languages can > only be executed via the freeswitch console. I was wondering whether it's > possible to run FreeSwitch at backend and have a piece of Java program > (outside the freeSwitch console) to invoke freeSwitch commands? > > Thanks, > -Jingwei > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/c9b67a7b/attachment.html From jingwei.yang at gmail.com Wed Jun 24 22:53:24 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 13:53:24 +0800 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> Message-ID: <13529f9d0906242253sa25bf71k237b6e68afed8c04@mail.gmail.com> Thanks Michael, fs_eslib sounds the one for Java. I'll give it a try. On Thu, Jun 25, 2009 at 1:38 PM, Michael Collins wrote: > FYI, > Any language that can establish a network socket and send/receive > information over that socket can be used to control FS. FreeSWITCH comes > with ESL - the event socket library - that can abstract away some of the > grunt work, but there isn't a Java one that I'm aware of. > -MC > > On Wed, Jun 24, 2009 at 7:52 PM, Jingwei Yang wrote: > >> Hi Folks, >> >> I understand freeSwitch is supporting a couple of languages for call >> controls like Lua, Javascript, Perl, Java... However, after digging into the >> detailed wiki pages, I found out the codes written in those languages can >> only be executed via the freeswitch console. I was wondering whether it's >> possible to run FreeSwitch at backend and have a piece of Java program >> (outside the freeSwitch console) to invoke freeSwitch commands? >> >> Thanks, >> -Jingwei >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/dc5ed2e0/attachment.html From peter.olsson at visionutveckling.se Thu Jun 25 00:36:47 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 25 Jun 2009 09:36:47 +0200 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d36d4319/attachment-0001.html From brian at freeswitch.org Thu Jun 25 01:15:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 03:15:39 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> Message-ID: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > I?ve been using FS as a gateway to our OCS server for some time. > It?s used just for testing, so it?s not really used every day. I > don?t know when, but after some trunk update (right now I running > r13945) of FS it doesn?t send the SIP traffic using tcp anymore (OCS > only accepts tcp or tls). > > My configuration is quite easy, I have a sofia gateway configured to > OCS, this has the parameter value="tcp"/> set in the config (nothing in the config has changed > for ages). Then in the dialplan I use this gateway to connect the > calls. When doing a siptrace I can see that the headers has > transport=tcp set correctly, but according to the trace it?s sent > using udp instead of tcp. > > Has something changed so I need to configure it in another way, or > is it just simply a bug? I just wanted to check this before issuing > a jira case and providing more specific information and debug traces > etc. > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/f366e706/attachment.html From brian at freeswitch.org Thu Jun 25 01:20:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 03:20:04 -0500 Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking In-Reply-To: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> References: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> Message-ID: <85A736A7-AB9F-44D5-BB21-A20BB94B8A6A@freeswitch.org> check that s is nil. /b On Jun 24, 2009, at 8:12 PM, John Wehle wrote: > What's the recommended way to check if the session constructor was > successful (i.e. the number could be dialed)? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a3676691/attachment.html From darklion11 at yahoo.com Thu Jun 25 01:39:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 01:39:38 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24196467.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> <24181933.post@talk.nabble.com> <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> <24196467.post@talk.nabble.com> Message-ID: <24199253.post@talk.nabble.com> Thanks a lot it works for me... Edmar Cruz wrote: > > If then, what bridge i shall call to? > > Like this? > > > > > dujinfang wrote: >> >> put your extension in dialplan/public.xml instead of sip_profiles/ >> external/myprofile.xml >> >> >> On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: >> >>> >>> Ooops.. Sorry wrong spelling... Same issue >>> >>> Jason White-14 wrote: >>>> >>>> Edmar Cruz wrote: >>>> >>>>> Here is my dialplan on sip_profiles/external/myprofile.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>> >>>> The above should be $1 not @1 >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24199253.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Thu Jun 25 03:16:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 03:16:38 -0700 (PDT) Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? Message-ID: <24200644.post@talk.nabble.com> Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? Thanks -- View this message in context: http://www.nabble.com/How-to-change-database-of-freeswitch-cdr-to-MySQL--tp24200644p24200644.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From chris.chen2004 at gmail.com Thu Jun 25 06:02:59 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 09:02:59 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> Message-ID: <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Two questions for you: 1) Do you have extension "888" in your public context? 2)Can you put your internal Ip address of FS in rtp-ip instead of $${bind_server_ip} just to make sure it get the right IP? 3) is not really required at least for my working setup behind the NAT router. On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang wrote: > Hi Chris, thanks for your help. Here's my client.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: > >> Please provide your client.xml detail with confidential information >> crossout, I have gtalk client and server working properly behind the NAT. >> I should be able to help you. >> >> Chris >> >> >> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: >> >>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >>> audio this time no matter what value disable-rtp-auto-adjust is... >>> >>> >>> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >>> >>>> uncomment ext-rtp-ip >>>> >>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>>> >>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put >>>> "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>>> parameter has (true or false), local IP is able to hear the echo but >>>> external ones still have no audio. >>>> >>>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>>> >>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember the >>>>> exact var. >>>>> >>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>>> >>>>> Hi Guys, >>>>> >>>>> Here's my situation: >>>>> >>>>> The freeswitch server and my machine are behind the same LAN. If I >>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>>> >>>>> However, external calls have no sound at all no matter whether this >>>>> param is commented out or not. >>>>> >>>>> Please kindly let me know what other params to set to resolve this >>>>> issue. >>>>> >>>>> Thanks, >>>>> -Jingwei >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/f3645c9e/attachment-0001.html From chris.chen2004 at gmail.com Thu Jun 25 06:07:50 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 09:07:50 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> Message-ID: <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: > Please open a jira and attach sip traces of register and phone calls. > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > > I?ve been using FS as a gateway to our OCS server for some time. It?s used > just for testing, so it?s not really used every day. I don?t know when, but > after some trunk update (right now I running r13945) of FS it doesn?t send > the SIP traffic using tcp anymore (OCS only accepts tcp or tls). > > My configuration is quite easy, I have a sofia gateway configured to OCS, > this has the parameter set in > the config (nothing in the config has changed for ages). Then in the > dialplan I use this gateway to connect the calls. When doing a siptrace I > can see that the headers has transport=tcp set correctly, but according to > the trace it?s sent using udp instead of tcp. > > Has something changed so I need to configure it in another way, or is it > just simply a bug? I just wanted to check this before issuing a jira case > and providing more specific information and debug traces etc. > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/7233dfd6/attachment.html From anthony.minessale at gmail.com Thu Jun 25 06:18:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 08:18:30 -0500 Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking In-Reply-To: <85A736A7-AB9F-44D5-BB21-A20BB94B8A6A@freeswitch.org> References: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> <85A736A7-AB9F-44D5-BB21-A20BB94B8A6A@freeswitch.org> Message-ID: <191c3a030906250618t145492a9x4e8de5d923dcc7ae@mail.gmail.com> and that s.ready() is true On Thu, Jun 25, 2009 at 3:20 AM, Brian West wrote: > check that s is nil. > /b > > On Jun 24, 2009, at 8:12 PM, John Wehle wrote: > > What's the recommended way to check if the session constructor was > successful (i.e. the number could be dialed)? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6aa1a36d/attachment.html From brian at freeswitch.org Thu Jun 25 07:15:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 09:15:16 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> Message-ID: <1FA565E6-4BCC-4C02-93DB-CD3409119F41@freeswitch.org> Please open a jira and attach sip traces. /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > I am having the same issue, now the SIP trunk over TCP between FS > and Exchange 2007 UM just stops working, just stuck in a loop like > this: > > 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > > On my Exchange 2007 side nothing was changed which used to work fine > with FS > > Chris > > On Thu, Jun 25, 2009 at 4:15 AM, Brian West > wrote: > Please open a jira and attach sip traces of register and phone calls. > > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > >> I?ve been using FS as a gateway to our OCS server for some time. >> It?s used just for testing, so it?s not really used every day. I >> don?t know when, but after some trunk update (right now I running >> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >> (OCS only accepts tcp or tls). >> >> My configuration is quite easy, I have a sofia gateway configured >> to OCS, this has the parameter > value="tcp"/> set in the config (nothing in the config has changed >> for ages). Then in the dialplan I use this gateway to connect the >> calls. When doing a siptrace I can see that the headers has >> transport=tcp set correctly, but according to the trace it?s sent >> using udp instead of tcp. >> >> Has something changed so I need to configure it in another way, or >> is it just simply a bug? I just wanted to check this before issuing >> a jira case and providing more specific information and debug >> traces etc. >> >> /Peter >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/97a5716c/attachment-0001.html From dujinfang at gmail.com Thu Jun 25 08:16:33 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 25 Jun 2009 23:16:33 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Message-ID: > 3) is not > really required at least for my working setup behind the NAT router. > ok, this param is originally added for another problem http://jira.freeswitch.org/browse/MODENDP-198 . But I think it might be useful for this. From peter.olsson at visionutveckling.se Thu Jun 25 08:22:22 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 25 Jun 2009 17:22:22 +0200 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE4885C@cooper> Done, added as issue SFSIP-157. Regards, Peter Olsson Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a4333c332936913812693! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a20735ee/attachment.html From paul.degt at gmail.com Thu Jun 25 08:24:20 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 11:24:20 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <24200644.post@talk.nabble.com> References: <24200644.post@talk.nabble.com> Message-ID: <4A4396A4.2060701@gmail.com> You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: > Hi, > > > How can i change my freeswitch database instead of CSV file, I make it > mysql. Can you tell me how? > > > Thanks > From anthony.minessale at gmail.com Thu Jun 25 08:31:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 10:31:48 -0500 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Message-ID: <191c3a030906250831m15208063j6314a6be5f51d80@mail.gmail.com> if you are behind nat , you will not want to disable auto-adjust that is what the feature is there to help you with. On Thu, Jun 25, 2009 at 10:16 AM, Seven Du wrote: > > 3) is not > > really required at least for my working setup behind the NAT router. > > > > ok, this param is originally added for another problem > http://jira.freeswitch.org/browse/MODENDP-198 > . But I think it might be useful for this. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/b5154148/attachment.html From brian at freeswitch.org Thu Jun 25 08:41:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 10:41:05 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE4885C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE4885C@cooper> Message-ID: <1B1FC434-6B4A-4A54-A547-1486A99AD53B@freeswitch.org> Fixed revision 13948. /b On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote: > Done, added as issue SFSIP-157. > > Regards, > > Peter Olsson > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] F?r Brian West > Skickat: den 25 juni 2009 10:16 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp > anymore... > > Please open a jira and attach sip traces of register and phone calls. > > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > > > I?ve been using FS as a gateway to our OCS server for some time. > It?s used just for testing, so it?s not really used every day. I > don?t know when, but after some trunk update (right now I running > r13945) of FS it doesn?t send the SIP traffic using tcp anymore (OCS > only accepts tcp or tls). > > My configuration is quite easy, I have a sofia gateway configured to > OCS, this has the parameter value="tcp"/> set in the config (nothing in the config has changed > for ages). Then in the dialplan I use this gateway to connect the > calls. When doing a siptrace I can see that the headers has > transport=tcp set correctly, but according to the trace it?s sent > using udp instead of tcp. > > Has something changed so I need to configure it in another way, or > is it just simply a bug? I just wanted to check this before issuing > a jira case and providing more specific information and debug traces > etc. > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > !DSPAM:4a4333c332936913812693! > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/172b4300/attachment-0001.html From brian at freeswitch.org Thu Jun 25 08:49:40 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 10:49:40 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A4396A4.2060701@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> Message-ID: <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > You can use FS XML Curl - FS sends XML CDRs to a web server of your > choice, and there you do whatever you want with these CDRs, like store > in a database. > There are also pre-built solutions available, check here: > http://versafon.com/versafonweb/CommercialSupport.jsp > > Edmar Cruz wrote: >> Hi, >> >> >> How can i change my freeswitch database instead of CSV file, I >> make it >> mysql. Can you tell me how? >> >> >> Thanks From brian at freeswitch.org Thu Jun 25 09:06:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:06:59 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> Message-ID: <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > I am having the same issue, now the SIP trunk over TCP between FS > and Exchange 2007 UM just stops working, just stuck in a loop like > this: > > 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > > On my Exchange 2007 side nothing was changed which used to work fine > with FS > > Chris > > On Thu, Jun 25, 2009 at 4:15 AM, Brian West > wrote: > Please open a jira and attach sip traces of register and phone calls. > > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > >> I?ve been using FS as a gateway to our OCS server for some time. >> It?s used just for testing, so it?s not really used every day. I >> don?t know when, but after some trunk update (right now I running >> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >> (OCS only accepts tcp or tls). >> >> My configuration is quite easy, I have a sofia gateway configured >> to OCS, this has the parameter > value="tcp"/> set in the config (nothing in the config has changed >> for ages). Then in the dialplan I use this gateway to connect the >> calls. When doing a siptrace I can see that the headers has >> transport=tcp set correctly, but according to the trace it?s sent >> using udp instead of tcp. >> >> Has something changed so I need to configure it in another way, or >> is it just simply a bug? I just wanted to check this before issuing >> a jira case and providing more specific information and debug >> traces etc. >> >> /Peter >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/e21e898a/attachment.html From paul.degt at gmail.com Thu Jun 25 09:15:11 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 12:15:11 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> Message-ID: <4A43A28F.7020404@gmail.com> I am an employee, why? Brian West wrote: > Are you the owner of Versafon? > > /b > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of your >> choice, and there you do whatever you want with these CDRs, like store >> in a database. >> There are also pre-built solutions available, check here: >> http://versafon.com/versafonweb/CommercialSupport.jsp >> >> Edmar Cruz wrote: >> >>> Hi, >>> >>> >>> How can i change my freeswitch database instead of CSV file, I >>> make it >>> mysql. Can you tell me how? >>> >>> >>> Thanks >>> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chris.chen2004 at gmail.com Thu Jun 25 09:24:24 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 12:24:24 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> Message-ID: <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: > Can you verify that this is fixed... I think its related to the same > issue... > /b > > On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > > I am having the same issue, now the SIP trunk over TCP between FS and > Exchange 2007 UM just stops working, just stuck in a loop like this: > > 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > > On my Exchange 2007 side nothing was changed which used to work fine with > FS > > Chris > > On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: > >> Please open a jira and attach sip traces of register and phone calls. >> /b >> >> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >> >> I?ve been using FS as a gateway to our OCS server for some time. It?s >> used just for testing, so it?s not really used every day. I don?t know when, >> but after some trunk update (right now I running r13945) of FS it doesn?t >> send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). >> >> My configuration is quite easy, I have a sofia gateway configured to OCS, >> this has the parameter set in >> the config (nothing in the config has changed for ages). Then in the >> dialplan I use this gateway to connect the calls. When doing a siptrace I >> can see that the headers has transport=tcp set correctly, but according to >> the trace it?s sent using udp instead of tcp. >> >> Has something changed so I need to configure it in another way, or is it >> just simply a bug? I just wanted to check this before issuing a jira case >> and providing more specific information and debug traces etc. >> >> /Peter >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/9fe8a8a6/attachment-0001.html From anthony.minessale at gmail.com Thu Jun 25 09:27:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 11:27:53 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43A28F.7020404@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> Message-ID: <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt wrote: > I am an employee, why? > > Brian West wrote: > > Are you the owner of Versafon? > > > > /b > > > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > > > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of your > >> choice, and there you do whatever you want with these CDRs, like store > >> in a database. > >> There are also pre-built solutions available, check here: > >> http://versafon.com/versafonweb/CommercialSupport.jsp > >> > >> Edmar Cruz wrote: > >> > >>> Hi, > >>> > >>> > >>> How can i change my freeswitch database instead of CSV file, I > >>> make it > >>> mysql. Can you tell me how? > >>> > >>> > >>> Thanks > >>> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/cabaf5cf/attachment.html From peter.olsson at visionutveckling.se Thu Jun 25 09:28:47 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 25 Jun 2009 18:28:47 +0200 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E229@cooper> What can I say - you guys provide far much better (and quicker) support then any commersial solution :) Thanks for the help! /Peter ________________________________ Fr?n: Brian West Skickat: den 25 juni 2009 17:53 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Fixed revision 13948. /b On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote: Done, added as issue SFSIP-157. Regards, Peter Olsson Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I?ve been using FS as a gateway to our OCS server for some time. It?s used just for testing, so it?s not really used every day. I don?t know when, but after some trunk update (right now I running r13945) of FS it doesn?t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it?s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a439d8f32931361515932! From anthony.minessale at gmail.com Thu Jun 25 09:35:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 11:35:15 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> Message-ID: <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen wrote: > I upgraded to 13950, still the same, keeping the same loop like the console > log showing: > 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > > Chris > > > On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: > >> Can you verify that this is fixed... I think its related to the same >> issue... >> /b >> >> On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: >> >> I am having the same issue, now the SIP trunk over TCP between FS and >> Exchange 2007 UM just stops working, just stuck in a loop like this: >> >> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> >> On my Exchange 2007 side nothing was changed which used to work fine with >> FS >> >> Chris >> >> On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: >> >>> Please open a jira and attach sip traces of register and phone calls. >>> /b >>> >>> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >>> >>> I?ve been using FS as a gateway to our OCS server for some time. It?s >>> used just for testing, so it?s not really used every day. I don?t know when, >>> but after some trunk update (right now I running r13945) of FS it doesn?t >>> send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). >>> >>> My configuration is quite easy, I have a sofia gateway configured to OCS, >>> this has the parameter set in >>> the config (nothing in the config has changed for ages). Then in the >>> dialplan I use this gateway to connect the calls. When doing a siptrace I >>> can see that the headers has transport=tcp set correctly, but according to >>> the trace it?s sent using udp instead of tcp. >>> >>> Has something changed so I need to configure it in another way, or is it >>> just simply a bug? I just wanted to check this before issuing a jira case >>> and providing more specific information and debug traces etc. >>> >>> /Peter >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d54749a6/attachment-0001.html From harmeet at litatel.com Thu Jun 25 09:35:49 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Thu, 25 Jun 2009 12:35:49 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: I didn't see the last message from Paul as advertizement. He was explaining about the FS XML Curl. I think he may have mentioned the link to versafon even if he didn't work there. On Thu, Jun 25, 2009 at 12:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > since you are advertising on our site regularly now perhaps you could ask > your boss to sponsor the project. > > > > On Thu, Jun 25, 2009 at 11:15 AM, paul.degt wrote: > >> I am an employee, why? >> >> Brian West wrote: >> > Are you the owner of Versafon? >> > >> > /b >> > >> > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: >> > >> > >> >> You can use FS XML Curl - FS sends XML CDRs to a web server of your >> >> choice, and there you do whatever you want with these CDRs, like store >> >> in a database. >> >> There are also pre-built solutions available, check here: >> >> http://versafon.com/versafonweb/CommercialSupport.jsp >> >> >> >> Edmar Cruz wrote: >> >> >> >>> Hi, >> >>> >> >>> >> >>> How can i change my freeswitch database instead of CSV file, I >> >>> make it >> >>> mysql. Can you tell me how? >> >>> >> >>> >> >>> Thanks >> >>> >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6fc0b914/attachment.html From brian at freeswitch.org Thu Jun 25 09:35:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:35:22 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> Message-ID: <8B617782-8E40-42F4-B132-6C1CA5B98137@freeswitch.org> I see what is wrong you're running it all on the same IP and the 302 is back to the same IP but a different port I need to fix that logic to compare the port number also. /b On Jun 25, 2009, at 11:24 AM, Chris Chen wrote: > I upgraded to 13950, still the same, keeping the same loop like the > console log showing: > 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > > Chris > > On Thu, Jun 25, 2009 at 12:06 PM, Brian West > wrote: > Can you verify that this is fixed... I think its related to the same > issue... > > /b > > On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > >> I am having the same issue, now the SIP trunk over TCP between FS >> and Exchange 2007 UM just stops working, just stuck in a loop like >> this: >> >> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> >> On my Exchange 2007 side nothing was changed which used to work >> fine with FS >> >> Chris >> >> On Thu, Jun 25, 2009 at 4:15 AM, Brian West >> wrote: >> Please open a jira and attach sip traces of register and phone calls. >> >> /b >> >> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >> >>> I?ve been using FS as a gateway to our OCS server for some time. >>> It?s used just for testing, so it?s not really used every day. I >>> don?t know when, but after some trunk update (right now I running >>> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >>> (OCS only accepts tcp or tls). >>> >>> My configuration is quite easy, I have a sofia gateway configured >>> to OCS, this has the parameter >> value="tcp"/> set in the config (nothing in the config has changed >>> for ages). Then in the dialplan I use this gateway to connect the >>> calls. When doing a siptrace I can see that the headers has >>> transport=tcp set correctly, but according to the trace it?s sent >>> using udp instead of tcp. >>> >>> Has something changed so I need to configure it in another way, or >>> is it just simply a bug? I just wanted to check this before >>> issuing a jira case and providing more specific information and >>> debug traces etc. >>> >>> /Peter >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/89d10c10/attachment-0001.html From msc at freeswitch.org Thu Jun 25 09:45:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 25 Jun 2009 09:45:25 -0700 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <13529f9d0906242253sa25bf71k237b6e68afed8c04@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> <13529f9d0906242253sa25bf71k237b6e68afed8c04@mail.gmail.com> Message-ID: <87f2f3b90906250945o2fa84e03t9f92fbede192dce2@mail.gmail.com> On Wed, Jun 24, 2009 at 10:53 PM, Jingwei Yang wrote: > Thanks Michael, fs_eslib sounds the one for Java. I'll give it a try. > Let us know how it goes. We like success stories! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/9bbf4d49/attachment.html From edpimentl at gmail.com Thu Jun 25 09:48:32 2009 From: edpimentl at gmail.com (EdPimentl) Date: Thu, 25 Jun 2009 12:48:32 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitch Sincerely, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/57290535/attachment.html From brian at freeswitch.org Thu Jun 25 09:52:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:52:09 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> Message-ID: <0E44A2AB-8232-401F-90FA-B0A87E3568FE@freeswitch.org> And possibly present the word FreeSWITCH in the proper case! ;) /b On Jun 25, 2009, at 11:48 AM, EdPimentl wrote: > To be fair ... > when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp > a common courtesy would be to provide a link to Freeswitch > > Sincerely, > -E Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d07b3692/attachment.html From brian at freeswitch.org Thu Jun 25 09:53:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:53:02 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> Message-ID: I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: > are you redirecting it to yourself by any chance because of some > proxy in your network? > > > On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen > wrote: > I upgraded to 13950, still the same, keeping the same loop like the > console log showing: > 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > > Chris > > > On Thu, Jun 25, 2009 at 12:06 PM, Brian West > wrote: > Can you verify that this is fixed... I think its related to the same > issue... > > /b > > On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > >> I am having the same issue, now the SIP trunk over TCP between FS >> and Exchange 2007 UM just stops working, just stuck in a loop like >> this: >> >> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> >> On my Exchange 2007 side nothing was changed which used to work >> fine with FS >> >> Chris >> >> On Thu, Jun 25, 2009 at 4:15 AM, Brian West >> wrote: >> Please open a jira and attach sip traces of register and phone calls. >> >> /b >> >> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >> >>> I?ve been using FS as a gateway to our OCS server for some time. >>> It?s used just for testing, so it?s not really used every day. I >>> don?t know when, but after some trunk update (right now I running >>> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >>> (OCS only accepts tcp or tls). >>> >>> My configuration is quite easy, I have a sofia gateway configured >>> to OCS, this has the parameter >> value="tcp"/> set in the config (nothing in the config has changed >>> for ages). Then in the dialplan I use this gateway to connect the >>> calls. When doing a siptrace I can see that the headers has >>> transport=tcp set correctly, but according to the trace it?s sent >>> using udp instead of tcp. >>> >>> Has something changed so I need to configure it in another way, or >>> is it just simply a bug? I just wanted to check this before >>> issuing a jira case and providing more specific information and >>> debug traces etc. >>> >>> /Peter >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/bd7e64df/attachment-0001.html From anthony.minessale at gmail.com Thu Jun 25 09:54:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 11:54:07 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: <191c3a030906250954j671a92d4p50ccdc83dad89485@mail.gmail.com> It's a suggestion not a demand. If you follow the link it's a list of products and services related to FS. We prefer that anyone who wants to sell stuff made from FS or as an accessory to FS would consider sponsoring the project or ClueCon. We provide both commercial and free support for FreeSWITCH and the amount of free help we have time to give is directly impacted by how many people who use FreeSWITCH for commercial purposes give back to us in the form of volunteer developers, support contracts and sponsorship. Trust me, if we don't ask very few will realize it on their own. On Thu, Jun 25, 2009 at 11:35 AM, Harmeet Singh wrote: > I didn't see the last message from Paul as advertizement. He was explaining > about the FS XML Curl. I think he may have mentioned the link to versafon > even if he didn't work there. > > > On Thu, Jun 25, 2009 at 12:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> since you are advertising on our site regularly now perhaps you could ask >> your boss to sponsor the project. >> >> >> >> On Thu, Jun 25, 2009 at 11:15 AM, paul.degt wrote: >> >>> I am an employee, why? >>> >>> Brian West wrote: >>> > Are you the owner of Versafon? >>> > >>> > /b >>> > >>> > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: >>> > >>> > >>> >> You can use FS XML Curl - FS sends XML CDRs to a web server of your >>> >> choice, and there you do whatever you want with these CDRs, like store >>> >> in a database. >>> >> There are also pre-built solutions available, check here: >>> >> http://versafon.com/versafonweb/CommercialSupport.jsp >>> >> >>> >> Edmar Cruz wrote: >>> >> >>> >>> Hi, >>> >>> >>> >>> >>> >>> How can i change my freeswitch database instead of CSV file, I >>> >>> make it >>> >>> mysql. Can you tell me how? >>> >>> >>> >>> >>> >>> Thanks >>> >>> >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6afeacd0/attachment.html From msc at freeswitch.org Thu Jun 25 09:54:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 25 Jun 2009 09:54:35 -0700 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> Message-ID: <87f2f3b90906250954o7ef9da6dp48fae6a140fc97aa@mail.gmail.com> On Thu, Jun 25, 2009 at 9:48 AM, EdPimentl wrote: > To be fair ... > when mentioning Freeswitch here > http://versafon.com/versafonweb/Software.jsp > a common courtesy would be to provide a link to Freeswitch > Not only that but spelling "FreeSWITCH" correctly would be a nice touch, no? :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/98b87a47/attachment.html From paul.degt at gmail.com Thu Jun 25 09:57:29 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 12:57:29 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: <4A43AC79.6080205@gmail.com> I talked about it but with the current state of VOIP market it's seems problematic. :-( I did not see my posts as strictly ads since we offer free software as well and also share my limited knowledge about FS. Anthony Minessale wrote: > since you are advertising on our site regularly now perhaps you could > ask your boss to sponsor the project. > > > On Thu, Jun 25, 2009 at 11:15 AM, paul.degt > wrote: > > I am an employee, why? > > Brian West wrote: > > Are you the owner of Versafon? > > > > /b > > > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > > > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of > your > >> choice, and there you do whatever you want with these CDRs, > like store > >> in a database. > >> There are also pre-built solutions available, check here: > >> http://versafon.com/versafonweb/CommercialSupport.jsp > >> > >> Edmar Cruz wrote: > >> > >>> Hi, > >>> > >>> > >>> How can i change my freeswitch database instead of CSV file, I > >>> make it > >>> mysql. Can you tell me how? > >>> > >>> > >>> Thanks > >>> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From paul.degt at gmail.com Thu Jun 25 10:02:37 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 13:02:37 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <0E44A2AB-8232-401F-90FA-B0A87E3568FE@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <0E44A2AB-8232-401F-90FA-B0A87E3568FE@freeswitch.org> Message-ID: <4A43ADAD.4040504@gmail.com> Done. :-) Brian West wrote: > And possibly present the word FreeSWITCH in the proper case! ;) > > /b > > On Jun 25, 2009, at 11:48 AM, EdPimentl wrote: > >> To be fair ... >> when mentioning Freeswitch >> here http://versafon.com/versafonweb/Software.jsp >> a common courtesy would be to provide a link to Freeswitch >> >> >> Sincerely, >> -E > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From paul.degt at gmail.com Thu Jun 25 10:04:18 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 13:04:18 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> Message-ID: <4A43AE12.70500@gmail.com> I will have to ask my boss about that, most probably he will ask same in return. EdPimentl wrote: > To be fair ... > when mentioning Freeswitch here > http://versafon.com/versafonweb/Software.jsp > a common courtesy would be to provide a link to Freeswitch > > > Sincerely, > -E > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Thu Jun 25 10:05:12 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 13:05:12 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> Message-ID: <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is working. Thanks for your great work. Chris On Thu, Jun 25, 2009 at 12:53 PM, Brian West wrote: > I found the problem... the fs_path refactor regression number 2 was just > fixed.. It was assuming the route_uri was the contact and making it stick to > the wrong place to send the invite... you should be able to update now and > it work correctly. > /b > > On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: > > are you redirecting it to yourself by any chance because of some proxy in > your network? > > > On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen wrote: > >> I upgraded to 13950, still the same, keeping the same loop like the >> console log showing: >> 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> >> Chris >> >> >> On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: >> >>> Can you verify that this is fixed... I think its related to the same >>> issue... >>> /b >>> >>> On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: >>> >>> I am having the same issue, now the SIP trunk over TCP between FS and >>> Exchange 2007 UM just stops working, just stuck in a loop like this: >>> >>> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> >>> On my Exchange 2007 side nothing was changed which used to work fine with >>> FS >>> >>> Chris >>> >>> On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: >>> >>>> Please open a jira and attach sip traces of register and phone calls. >>>> /b >>>> >>>> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >>>> >>>> I?ve been using FS as a gateway to our OCS server for some time. It?s >>>> used just for testing, so it?s not really used every day. I don?t know when, >>>> but after some trunk update (right now I running r13945) of FS it doesn?t >>>> send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). >>>> >>>> My configuration is quite easy, I have a sofia gateway configured to >>>> OCS, this has the parameter >>>> set in the config (nothing in the config has changed for ages). Then in the >>>> dialplan I use this gateway to connect the calls. When doing a siptrace I >>>> can see that the headers has transport=tcp set correctly, but according to >>>> the trace it?s sent using udp instead of tcp. >>>> >>>> Has something changed so I need to configure it in another way, or is it >>>> just simply a bug? I just wanted to check this before issuing a jira case >>>> and providing more specific information and debug traces etc. >>>> >>>> /Peter >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/96db9043/attachment-0001.html From intralanman at freeswitch.org Thu Jun 25 10:08:30 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 25 Jun 2009 13:08:30 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AE12.70500@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com> Message-ID: <4A43AF0E.6010809@freeswitch.org> On 06/25/2009 01:04 PM, paul.degt wrote: > I will have to ask my boss about that, most probably he will ask same in > That doesn't really make sense... FreeSWITCH isn't using or benefitting from your software... but yours is from FreeSWITCH -Ray From anthony.minessale at gmail.com Thu Jun 25 10:09:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 12:09:08 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AC79.6080205@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <4A43AC79.6080205@gmail.com> Message-ID: <191c3a030906251009n39c9dcb7j1b837a4ce84ca858@mail.gmail.com> Right which is why i said it was a "suggestion" not a demand. If you want to help answer email in here or answer questions on a regular basis that's just as valuable. if your boss has no money to sponsor maybe he can donate you =D On Thu, Jun 25, 2009 at 11:57 AM, paul.degt wrote: > I talked about it but with the current state of VOIP market it's seems > problematic. :-( > I did not see my posts as strictly ads since we offer free software as > well and also share my limited knowledge about FS. > > Anthony Minessale wrote: > > since you are advertising on our site regularly now perhaps you could > > ask your boss to sponsor the project. > > > > > > On Thu, Jun 25, 2009 at 11:15 AM, paul.degt > > wrote: > > > > I am an employee, why? > > > > Brian West wrote: > > > Are you the owner of Versafon? > > > > > > /b > > > > > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > > > > > > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of > > your > > >> choice, and there you do whatever you want with these CDRs, > > like store > > >> in a database. > > >> There are also pre-built solutions available, check here: > > >> http://versafon.com/versafonweb/CommercialSupport.jsp > > >> > > >> Edmar Cruz wrote: > > >> > > >>> Hi, > > >>> > > >>> > > >>> How can i change my freeswitch database instead of CSV file, I > > >>> make it > > >>> mysql. Can you tell me how? > > >>> > > >>> > > >>> Thanks > > >>> > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/12d44479/attachment.html From anthony.minessale at gmail.com Thu Jun 25 10:18:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 12:18:02 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AF0E.6010809@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com> <4A43AF0E.6010809@freeswitch.org> Message-ID: <191c3a030906251018y2727ad3cgf04a89049334a75b@mail.gmail.com> FreeSWITCH Solutions will soon be offering a product gallery where companies who use FS can become certified partners and display their products. On Thu, Jun 25, 2009 at 12:08 PM, Raymond Chandler < intralanman at freeswitch.org> wrote: > On 06/25/2009 01:04 PM, paul.degt wrote: > > I will have to ask my boss about that, most probably he will ask same in > > > > That doesn't really make sense... FreeSWITCH isn't using or benefitting > from your software... but yours is from FreeSWITCH > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/cd917142/attachment.html From paul.degt at gmail.com Thu Jun 25 10:23:35 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 13:23:35 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AF0E.6010809@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com> <4A43AF0E.6010809@freeswitch.org> Message-ID: <4A43B297.7030005@gmail.com> FS users may well be benefiting - thus FS itself benefits as well indirectly. Raymond Chandler wrote: > On 06/25/2009 01:04 PM, paul.degt wrote: > >> I will have to ask my boss about that, most probably he will ask same in >> >> > > That doesn't really make sense... FreeSWITCH isn't using or benefitting > from your software... but yours is from FreeSWITCH > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From d at d-man.org Thu Jun 25 10:34:18 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 25 Jun 2009 10:34:18 -0700 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43B297.7030005@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com><4A43AF0E.6010809@freeswitch.org> <4A43B297.7030005@gmail.com> Message-ID: <70AABAE3FDEC46B1953D0DAA17F7DE2E@test> Is it possible to unsusbcribe from specific threads on this list? Specifically I am looking for C code that removes useless banter so my brain doesn't hurt so much... -----Original Message----- From: paul.degt [mailto:paul.degt at gmail.com] Sent: Thursday, June 25, 2009 10:24 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? FS users may well be benefiting - thus FS itself benefits as well indirectly. Raymond Chandler wrote: > On 06/25/2009 01:04 PM, paul.degt wrote: > >> I will have to ask my boss about that, most probably he will ask same >> in >> >> > > That doesn't really make sense... FreeSWITCH isn't using or > benefitting from your software... but yours is from FreeSWITCH > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From d at d-man.org Thu Jun 25 10:35:29 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 25 Jun 2009 10:35:29 -0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> Message-ID: Did this work? Would love an update on this error/issue. _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, June 24, 2009 8:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway try adding before the bridge and report back results. Mike On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: Dear All, Look like nibblebill does't work with multiple gatreway. I try > before the bridge and report back results. > > Mike > > On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: > > Dear All, > > Look like nibblebill does't work with multiple gatreway. > I try > data="nibble_account=0838833133"/> > > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3] > sofia/external/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5] > sofia/external/6626734000 at 202.xxx.xxx.xxx> > > nibblebill not found nibble_rate > > But > > data="nibble_account=0838833133"/> > > session.execute("set", "import=nibble_account"); > session.execute("bridge", "{absolute_codec_string='GSM,G729'} > [nibble_rate=0.5,nibble_account=0838833133]sofia/external/xxxx at xxxx.xxx.xxx.xx > "); > > when call connected nibble do nothing i found heartbeat > > mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! > when call disconnect nibble update amont. > mod_nibblebill.c:478 Billing 16 secs > > I think nibble still not found variable channel. > > Let's me share more information > > I want to use nibblebill for callingcard. (i have develop billing by > myself). i plan to use javascript connect to ODBC > when customer call my script query balance and say. > and then i loop for get destination (my customer want to dial many > number). when i got number my script query > gateway from DB. i have 3 route and order by cost. > First plan i use > session.execute("bridge", > "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/ > xxxx at provder1| > [nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/ > xxxx at provder2"); > i modify nibblebill for match provider with my billing. > this case still fail. > > now i try > > if (session.ready()){ > s = new Session("{absolute_codec_string='GSM,G729'}sofia/ > external/xxx at provider1" > } > if (s.ready()){ > session.execute("set", "nibble_rate=2.5"); > session.execute("set", "nibble_account="+acaller); > session.execute("set", "hangup_after_bridge=false"); > session.execute("set", "provider_id="+dialprovider_id[1]); > bridge(session,s); > } > > and check hangup cause before try other provider. > > > > Please guide me it's right way or not ? > > > Dome C. > > > 2009/6/26 Darren Schreiber > Did this work? Would love an update on this error/issue. > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Wednesday, June 24, 2009 8:15 AM > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway > > try adding > > before the bridge and report back results. > > Mike > > On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: > >> Dear All, >> >> Look like nibblebill does't work with multiple gatreway. >> I try >> >> > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx >> |[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> nibblebill not found nibble_rate >> >> But >> >> >> > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx >> |sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> Work fine >> >> What's difference from set application and [] ? >> >> Best Regards. >> Dome C. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/55a9003a/attachment-0001.html From john at feith.com Thu Jun 25 12:35:55 2009 From: john at feith.com (John Wehle) Date: Thu, 25 Jun 2009 15:35:55 -0400 (EDT) Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking Message-ID: <200906251935.n5PJZtUt004125@jwlab.FEITH.COM> >> What's the recommended way to check if the session constructor was >> successful (i.e. the number could be dialed)? > check that s is nil. Doesn't work ... s is never nil. Type shows it as userdata even if Session failed. Specifically my test was: local s = freeswitch.Session ( "{ignore_early_media=true,origination_caller_id_name=" .. caller .. "}loopback/" .. destination .. "/default/XML") stream:write (type(s)) if s == nil then stream:write ("-ERR call failed\n") return end and I dialed an unreachable number. > and that s.ready() is true Checking s.ready() results in: [ERR] freeswitch_lua.cpp:102 session is not initalized if Session failed. What I'm looking for is a way to try to originate a call which doesn't throw ERR messages if the attempt fails. Explicitly calling session.originate seems to allow you to check if the call was successful ... is there a particular reason it's discouraged? I'm happy to avoid it if a better approach is available, however I'm having trouble finding one. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From anthony.minessale at gmail.com Thu Jun 25 12:50:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 14:50:30 -0500 Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking In-Reply-To: <200906251935.n5PJZtUt004125@jwlab.FEITH.COM> References: <200906251935.n5PJZtUt004125@jwlab.FEITH.COM> Message-ID: <191c3a030906251250k4050dba9sdef0a3c633898cb5@mail.gmail.com> I think this is an oversight update to trunk and session.ready() should work as expected. On Thu, Jun 25, 2009 at 2:35 PM, John Wehle wrote: > >> What's the recommended way to check if the session constructor was > >> successful (i.e. the number could be dialed)? > > > check that s is nil. > > Doesn't work ... s is never nil. Type shows it as userdata > even if Session failed. Specifically my test was: > > local s = freeswitch.Session ( > "{ignore_early_media=true,origination_caller_id_name=" .. > caller .. "}loopback/" .. destination .. "/default/XML") > > stream:write (type(s)) > > if s == nil then > stream:write ("-ERR call failed\n") > return > end > > and I dialed an unreachable number. > > > and that s.ready() is true > > Checking s.ready() results in: > > [ERR] freeswitch_lua.cpp:102 session is not initalized > > if Session failed. > > What I'm looking for is a way to try to originate a call which doesn't > throw ERR messages if the attempt fails. > > Explicitly calling session.originate seems to allow you to check if > the call was successful ... is there a particular reason it's discouraged? > > I'm happy to avoid it if a better approach is available, however I'm > having trouble finding one. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/3eeea664/attachment.html From vince.freeswitch at hightek.org Thu Jun 25 13:44:56 2009 From: vince.freeswitch at hightek.org (Vincent) Date: Thu, 25 Jun 2009 15:44:56 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> Message-ID: <20090625204456.GA45220@quark.hightek.org> On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > > > On Jun 23, 2009, at 10:15 PM, Vincent Stemen wrote: > > > Here is what I did and the results: > > > > ==================================================== > > Checked out the current trunk with svn. > > > > Patched /usr/include/sys/resource.h > > > > Since Dragonfly has fixed or will be fixing this future releases I > > patched the > > system header to add RLIMIT_AS rather than patching freeswitch to use > > RLIMIT_VMEM. > > Can we make a patch ifdefing on RLIMIT_AS to make this always work > without patches to system header files? Thanks for the responses Michael. I did this for attempting to compile freeswitch-1.0.3 and trunk as of a couple months ago. It would not apply to the current freeswitch trunk though. Apparently there have been changes to that area of the code. Since RLIMIT_AS is apparently a posix standard definition, I think this is fixed in Dragonfly HEAD and should not be a problem with future releases. I could go ahead and make a new patch when I get a chance if you still want me to, for compatibility with older Dragonfly releases. > > Compiling > > ========= > > > > Still lots of warnings of: > > warning: return makes pointer from integer without a cast > > > > Errors: > > It is apparently not checking return codes from make. It continues > > even when > > there are errors. Is this intentional?? > > > > su_alloc.c: In function `su_salloc': > > su_alloc.c:1518: warning: return makes pointer from integer without > > a cast > > gmake[9]: *** [su_alloc.lo] Error 1 > > gmake[8]: *** [all] Error 2 > > Making all in features > > LTCOMPILE features.lo > > ... > > > > Making all in sresolv > > LTCOMPILE sres.lo > > LTCOMPILE sres_cache.lo > > LTCOMPILE sres_blocking.lo > > LTCOMPILE sresolv.lo > > LTCOMPILE sres_sip.lo > > sres_sip.c: In function `sres_sip_new': > > sres_sip.c:267: warning: return makes pointer from integer without > > a cast > > gmake[8]: *** [sres_sip.lo] Error 1 > > Making all in ipt > > LTCOMPILE base64.lo > > LTCOMPILE token64.lo > > LINK libipt.la > > ... > > > > There are about 12 errors of this nature before ending with > > > > Making all in nua > > LTCOMPILE nua.lo > > nua.c: In function `nua_create': > > nua.c:141: warning: return makes pointer from integer without a cast > > nua.c:144: warning: return makes pointer from integer without a cast > > gmake[9]: *** [nua.lo] Error 1 > > gmake[8]: *** [all] Error 2 > > gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed > > by `libsofia-sip-ua.la'. Stop. > > gmake[7]: *** [all-recursive] Error 1 > > Making all in packages > > gmake[6]: *** [all-recursive] Error 1 > > gmake[5]: *** [all] Error 2 > > gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > > freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- > > ua.la] Error 2 > > gmake[3]: *** [mod_sofia-all] Error 1 > > gmake[2]: *** [all-recursive] Error 1 > > Making all in build > > +-------- FreeSWITCH Build Complete -----------+ > > + FreeSWITCH has been successfully built. + > > + Install by running: + > > + + > > + gmake install + > > +----------------------------------------------+ > > gmake[1]: *** [all-recursive] Error 1 > > gmake: *** [all] Error 2 > > > > Can you post a bug to Jira.freeswitch.org with all these warnings, > even better with patches to fix it. > > > > > It says it has been successfully built. Apparently part of the same > > problem of > > not checking the return codes. > > > > Patches to fix this appreciated Heh :-) OK. If I get it working and we end up using freeswitch, I will probably take a look at seeing if I can fix some or all of these warnings and create patches. > > It does not say what most of the errors are except for near the last > > when it > > says > > No rule to make target `iptsec/libiptsec.la' > > > > It just says "Error 1" or Error 2" which does not tell me what the > > problem is. From vince.freeswitch at hightek.org Thu Jun 25 14:49:10 2009 From: vince.freeswitch at hightek.org (Vincent) Date: Thu, 25 Jun 2009 16:49:10 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090624031950.GD2623@hijacked.us> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <20090624031950.GD2623@hijacked.us> Message-ID: <20090625214910.GB45220@quark.hightek.org> On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote: > On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: > > Ok. I did this. > > > > Compilation still failed but there are significant improvements since > > the last time. > > > > Here is what I did and the results: > > > > It looks like some the games that sofia plays with errno makes Dragonfly > unhappy. I also noticed that where the code checks for BSD-like systems > (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is > omitted, so obviously one of the first steps would be to fix that (if > applicable). > > If you disable mod_sofia in modules conf, do the rest of the default > modules build OK? OK. I commented out endpoints/mod_sofia. It looks like that eliminated all the errors except the one I get at the end. making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/nsprpub/dist/include/nspr/.: File exists gmake[9]: *** [export] Error 1 gmake[8]: *** [export] Error 2 gmake[7]: *** [export] Error 2 gmake[6]: *** [export] Error 2 gmake[5]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 So, it looks like most all the problems, except for that symlink error, including the pointer cast warnings, are related to the sofia module. I notice a lot of the modules seem to be redirecting the output somewhere. Not only do they just say Error 1 or Error 2 when there is an error, they also do not show the compile commands. They just output something like "Making built-sources in su" or "Compiling src/switch_apr.c ...". Is there a log file somewhere that contains the actual compile commands and error output so you can find out what happened when there is a error? Or perhaps a configuration to enable it to come out on the console? > For the record, DragonFly and FreeBSD have rather seriously diverged at > this point, DragonFly forked from FreeBSD back in the 4.10 days or so > and has changed a *lot* of things since, so I don't think it's gonna be > quite as easy as you expected (but it's far from impossible either). > > Andrew True, architecturally Dragonfly is becoming very different. They seem to be trying to maintain fairly good API compatibility though. Enough to constantly allow them to bring across major sub-systems, such as sound and SATA drivers, etc, from FreeBSD. So far, they have been pretty good about correcting it as soon as possible whenever one of us finds an incompatibility (Such as the RLIMIT_AS issue). Usually, all I have to do is add "-D__FreeBSD__" to CFLAGS and CPPFLAGS to compile packages that do not natively know about Dragonfly yet. Which is what I am doing with freeswitch. From drago at windstream.net Thu Jun 25 14:55:28 2009 From: drago at windstream.net (Drago Totev) Date: Thu, 25 Jun 2009 17:55:28 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> Message-ID: <000001c9f5df$a7823410$f6869c30$@net> Will it be a Windows build with the fix available soon? Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Chen Sent: Thursday, June 25, 2009 1:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is working. Thanks for your great work. Chris On Thu, Jun 25, 2009 at 12:53 PM, Brian West wrote: I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen wrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/637e70f7/attachment-0001.html From brian at freeswitch.org Thu Jun 25 14:57:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 16:57:48 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <000001c9f5df$a7823410$f6869c30$@net> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> <000001c9f5df$a7823410$f6869c30$@net> Message-ID: I don't think the windows build was updated to include the bug... but you can build it with MSVC Express Edition which is Free from Microsoft. /b On Jun 25, 2009, at 4:55 PM, Drago Totev wrote: > Will it be a Windows build with the fix available soon? > > Drago Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/08834a2f/attachment.html From mike at jerris.com Thu Jun 25 15:06:04 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 25 Jun 2009 18:06:04 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090625214910.GB45220@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <20090624031950.GD2623@hijacked.us> <20090625214910.GB45220@quark.hightek.org> Message-ID: <9E2119E8-99A3-40B9-960F-539B881CF1EE@jerris.com> On Jun 25, 2009, at 5:49 PM, Vincent wrote: > On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote: >> On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: >>> Ok. I did this. >>> >>> Compilation still failed but there are significant improvements >>> since >>> the last time. >>> >>> Here is what I did and the results: >>> >> >> It looks like some the games that sofia plays with errno makes >> Dragonfly >> unhappy. I also noticed that where the code checks for BSD-like >> systems >> (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, >> DragonFly is >> omitted, so obviously one of the first steps would be to fix that (if >> applicable). >> >> If you disable mod_sofia in modules conf, do the rest of the default >> modules build OK? > > OK. I commented out endpoints/mod_sofia. It looks like that > eliminated > all the errors except the one I get at the end. > > making all mod_spidermonkey > cd config; gmake -j1 export > cd pr; gmake -j1 export > cd include; gmake export > cd md; gmake export > ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ > ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/nsprpub/ > dist/include/nspr/.: File exists > gmake[9]: *** [export] Error 1 > gmake[8]: *** [export] Error 2 > gmake[7]: *** [export] Error 2 > gmake[6]: *** [export] Error 2 > gmake[5]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > freeswitch-20090623/libs/js/libjs.la] Error 2 > gmake[4]: *** [all] Error 1 > gmake[3]: *** [mod_spidermonkey-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 you can also comment out that module and see if you get further. > > > So, it looks like most all the problems, except for that symlink > error, > including the pointer cast warnings, are related to the sofia module. > > I notice a lot of the modules seem to be redirecting the output > somewhere. > > Not only do they just say Error 1 or Error 2 when there is an error, > they > also do not show the compile commands. They just output something > like > "Making built-sources in su" or "Compiling src/switch_apr.c ...". Is > there a log file somewhere that contains the actual compile commands > and > error output so you can find out what happened when there is a error? > Or perhaps a configuration to enable it to come out on the console? VERBOSE=1 gmake >> For the record, DragonFly and FreeBSD have rather seriously >> diverged at >> this point, DragonFly forked from FreeBSD back in the 4.10 days or so >> and has changed a *lot* of things since, so I don't think it's >> gonna be >> quite as easy as you expected (but it's far from impossible either). >> >> Andrew > > True, architecturally Dragonfly is becoming very different. They seem > to be trying to maintain fairly good API compatibility though. Enough > to constantly allow them to bring across major sub-systems, such as > sound and SATA drivers, etc, from FreeBSD. So far, they have been > pretty good about correcting it as soon as possible whenever one of us > finds an incompatibility (Such as the RLIMIT_AS issue). > > Usually, all I have to do is add "-D__FreeBSD__" to CFLAGS and > CPPFLAGS > to compile packages that do not natively know about Dragonfly yet. > Which is what I am doing with freeswitch. > From Richard.Lamkin at mettoni.com Thu Jun 25 15:08:29 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 25 Jun 2009 23:08:29 +0100 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7CC9@nickel.mettonigroup.com> Dear Anthony and Brian, Firstly please accept my apologies for wasting your time. Brian's request for the SVN number prompted me to realise I was running with an out of date version of FS. When I synced up to the head of the trunk and reran my tests the scenario I described below worked perfectly with no stuck calls. Therefore the sequence Park, Ringing (ring back), Redirect using the event API has provided me with the automated redirection I was seeking. Thank you for your advice earlier this week and prompt turnaround of fixes for the problems I encountered with bridged and deflected calls. Regards Richard Lamkin Richard.lamkin at mettoni.com From: Brian West [mailto:brian at freeswitch.org] Sent: 24 June 2009 19:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls left on FS after redirect offof FS I have tried to reproduce this issue but haven't been able too... What SVN Rev are you on? /b On Jun 24, 2009, at 10:29 AM, Richard Lamkin wrote: I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg call-command: execute execute-app-name: redirect execute-app-arg: sip:@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg call-command: execute execute-app-name: hangup execute-app-arg: Using command "api show channels" I find the following held on FS The only way I've found to remove these calls is "api hupall" ------------------------- uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. ------------------- The SIP signalling is correct with an outgoing "302 moved temporarily" [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either "api hupall", or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to "api hupall". "api hupall" kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lamkin at mettoni.com ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/bc93458a/attachment-0001.html From brian at freeswitch.org Thu Jun 25 16:56:14 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 18:56:14 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <24A1F18D-71EE-4751-967E-A450ED9185FF@freeswitch.org> References: <1384.1245405662@enterux.com> <24A1F18D-71EE-4751-967E-A450ED9185FF@freeswitch.org> Message-ID: <8B2560DD-4788-4823-A121-1C9A7C4C159A@freeswitch.org> Everyone that wanted to help this project to pay Arsen to write this please paypal brian at freeswitch.org when you can so I can gather it all up and send it to Arsen... Everyone that has sent money already thank you... ;) http://code.google.com/p/unimrcp/source/browse/trunk/plugins/mrcp-flite/src/mrcp_flite.c http://code.google.com/p/unimrcp/source/browse/trunk/plugins/mrcp-pocketsphinx/src/mrcp_pocketsphinx.c So the progress is moving forward.... Please pitch in. Thanks, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6c8b9eb8/attachment.html From danishmoosa at gmail.com Thu Jun 25 17:34:52 2009 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Fri, 26 Jun 2009 06:34:52 +0600 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? Message-ID: Hi Can somebody tell if FS freezes/crashes due to any reason. Does it logs both log of each call before dying? If we run it on large scale like 2-3k calls , a simple crash can cost a lot if it dies silently. One more aspect is , after freezing it will no more send/rec packets to any endpoint ,may result in inaccurate logging on endpoint. It should somehow send BYE ? -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/c3ee4b28/attachment.html From brian at freeswitch.org Thu Jun 25 17:44:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 19:44:20 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: Message-ID: No the goal is to NOT crash in the first place. Are you experiencing a crash? If so http://wiki.freeswitch.org/wiki/Reporting_Bugs is how you would report it. Thanks, Brian On Jun 25, 2009, at 7:34 PM, Muhammad Danish Moosa wrote: > Hi > > Can somebody tell if FS freezes/crashes due to any reason. Does it > logs both log of each call before dying? > > If we run it on large scale like 2-3k calls , a simple crash can > cost a lot if it dies silently. One more aspect is , after freezing > it will no more send/rec packets to any endpoint ,may result in > inaccurate logging on endpoint. It should somehow send BYE ? > > -- > Muhammad Danish Moosa Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/1e03c0c3/attachment.html From mgg at giagnocavo.net Thu Jun 25 18:16:13 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 25 Jun 2009 21:16:13 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> Well this isn't specific to FS crashing. The machine losing power would have the same effect, no? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Danish Moosa Sent: Thursday, June 25, 2009 6:35 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] CDR loss possibility if FS freezes? Hi Can somebody tell if FS freezes/crashes due to any reason. Does it logs both log of each call before dying? If we run it on large scale like 2-3k calls , a simple crash can cost a lot if it dies silently. One more aspect is , after freezing it will no more send/rec packets to any endpoint ,may result in inaccurate logging on endpoint. It should somehow send BYE ? -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/398e914a/attachment.html From jingwei.yang at gmail.com Thu Jun 25 19:15:28 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 26 Jun 2009 10:15:28 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Message-ID: <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> Hi Chris. thanks for the reply. Here're my answers. On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen wrote: > Two questions for you: > > 1) Do you have extension "888" in your public context? What public context are you saying? I only defined 888.xml in /usr/local/freeswitch/conf/directory/default. > 2)Can you put your internal Ip address of FS in rtp-ip instead of > $${bind_server_ip} just to make sure it get the right IP? I changed it to the internal Ip, but still no echo. > > 3) is not really > required at least for my working setup behind the NAT router. Thanks, I've commented it out. > > > > On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang wrote: > >> Hi Chris, thanks for your help. Here's my client.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: >> >>> Please provide your client.xml detail with confidential information >>> crossout, I have gtalk client and server working properly behind the NAT. >>> I should be able to help you. >>> >>> Chris >>> >>> >>> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: >>> >>>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >>>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >>>> audio this time no matter what value disable-rtp-auto-adjust is... >>>> >>>> >>>> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >>>> >>>>> uncomment ext-rtp-ip >>>>> >>>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>>>> >>>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and >>>>> put "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>>>> parameter has (true or false), local IP is able to hear the echo but >>>>> external ones still have no audio. >>>>> >>>>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>>>> >>>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember >>>>>> the exact var. >>>>>> >>>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>>>> >>>>>> Hi Guys, >>>>>> >>>>>> Here's my situation: >>>>>> >>>>>> The freeswitch server and my machine are behind the same LAN. If I >>>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>>>> >>>>>> However, external calls have no sound at all no matter whether this >>>>>> param is commented out or not. >>>>>> >>>>>> Please kindly let me know what other params to set to resolve this >>>>>> issue. >>>>>> >>>>>> Thanks, >>>>>> -Jingwei >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/0185e1fa/attachment-0001.html From chris.chen2004 at gmail.com Thu Jun 25 19:30:19 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 22:30:19 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> Message-ID: <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> I guess you have the problem here, in client.xml you have but you only define extension 888 in default context, that's why nobody can reach you from public. under /usr/local/freeswitch/conf/dialplan define extension 888 in public.xml to the proper extension you expect, and check the console log from fs_cli when you do gtalk calling to your gmail client, you will find out the solution to your issue. chris On Thu, Jun 25, 2009 at 10:15 PM, Jingwei Yang wrote: > Hi Chris. thanks for the reply. Here're my answers. > > On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen wrote: > >> Two questions for you: >> >> 1) Do you have extension "888" in your public context? > > > What public context are you saying? I only defined 888.xml in > /usr/local/freeswitch/conf/directory/default. > > >> 2)Can you put your internal Ip address of FS in rtp-ip instead of >> $${bind_server_ip} just to make sure it get the right IP? > > > I changed it to the internal Ip, but still no echo. > > >> >> 3) is not really >> required at least for my working setup behind the NAT router. > > > Thanks, I've commented it out. > > >> >> >> >> On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang wrote: >> >>> Hi Chris, thanks for your help. Here's my client.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: >>> >>>> Please provide your client.xml detail with confidential information >>>> crossout, I have gtalk client and server working properly behind the NAT. >>>> I should be able to help you. >>>> >>>> Chris >>>> >>>> >>>> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: >>>> >>>>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >>>>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >>>>> audio this time no matter what value disable-rtp-auto-adjust is... >>>>> >>>>> >>>>> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >>>>> >>>>>> uncomment ext-rtp-ip >>>>>> >>>>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>>>>> >>>>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and >>>>>> put "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>>>>> parameter has (true or false), local IP is able to hear the echo but >>>>>> external ones still have no audio. >>>>>> >>>>>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>>>>> >>>>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember >>>>>>> the exact var. >>>>>>> >>>>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>>>>> >>>>>>> Hi Guys, >>>>>>> >>>>>>> Here's my situation: >>>>>>> >>>>>>> The freeswitch server and my machine are behind the same LAN. If I >>>>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>>>>> >>>>>>> However, external calls have no sound at all no matter whether this >>>>>>> param is commented out or not. >>>>>>> >>>>>>> Please kindly let me know what other params to set to resolve this >>>>>>> issue. >>>>>>> >>>>>>> Thanks, >>>>>>> -Jingwei >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d747d0f1/attachment.html From dome at tel.co.th Thu Jun 25 19:38:45 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 09:38:45 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> <8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com> <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> Message-ID: <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> 2009/6/26 Michael Jerris : > I said to just add the set import=nibble_rate, your re-setting it for no > reason (and getting rid of the change that should have helped) by your > import=nibble_account line I test it agin. import work. nibble can see nibble_rate , nibble_account in channel but i can't change nibble heratbeat so nibble use default heartbeat. Dome C. > Mike > On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: > > Just test. > i use javascript > > ?????? session.execute("set", "import=nibble_rate"); > ?????? session.execute("set", "import=nibble_account"); > ?????? session.execute("bridge", > "{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=0838833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); > > when call connected nibble do nothing? i found heartbeat > > mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! > when call disconnect nibble update amont. > mod_nibblebill.c:478 Billing 16 secs > > I think nibble still not found variable channel. > > Let's me share more information > > I want to use nibblebill for callingcard. (i have develop billing by > myself). i plan to use javascript connect to ODBC > when customer call my script query balance and say. > and then i loop for get destination (my customer want to dial many number). > when i got number my script query > gateway from DB.? i have 3 route and order by cost. > First plan i use > session.execute("bridge", > "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/xxxx at provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/xxxx at provder2"); > i modify nibblebill for match provider with my billing. > this case still fail. > > now i try > > ??? if > (session.ready()){ > ??????? s = new > Session("{absolute_codec_string='GSM,G729'}sofia/external/xxx at provider1" > > } > ??? if > (s.ready()){ > ??????? session.execute("set", > "nibble_rate=2.5"); > ??????? session.execute("set", > "nibble_account="+acaller); > ??????? session.execute("set", > "hangup_after_bridge=false"); > ??????? session.execute("set", > "provider_id="+dialprovider_id[1]); > > bridge(session,s); > ??? } > > and check hangup cause before try other provider. > > > > Please guide me it's right way or not ? > > > Dome C. > > > 2009/6/26 Darren Schreiber >> >> Did this work? Would love an update on this error/issue. >> ________________________________ >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Wednesday, June 24, 2009 8:15 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >> >> try adding >> >> before the bridge and report back results. >> Mike >> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >> >> Dear All, >> >> Look like nibblebill does't work with multiple gatreway. >> I try >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> nibblebill not found nibble_rate >> >> But >> ??????? >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> Work fine >> >> What's difference from set application and []? ? >> >> Best Regards. >> Dome C. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Jun 25 19:40:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 21:40:04 -0500 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> Message-ID: I have been testing dingaling all day... I added switch_nat routines to poke holes in the nat if needed if you're behind upnp or nat-pmp. /b On Jun 25, 2009, at 9:30 PM, Chris Chen wrote: > I guess you have the problem here, > in client.xml you have > > > but you only define extension 888 in default context, > that's why nobody can reach you from public. > > under /usr/local/freeswitch/conf/dialplan > > define extension 888 in public.xml to the proper extension you > expect, and check the console log from fs_cli when you do gtalk > calling to your gmail client, you will find out the solution to your > issue. > > chris From brian at freeswitch.org Thu Jun 25 20:09:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:09:11 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> Message-ID: <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. /b On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: > Well this isn?t specific to FS crashing. The machine losing power > would have the same effect, no? > > -Michael > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/dbba1a79/attachment.html From shannon at sacredhearts.us Thu Jun 25 20:32:37 2009 From: shannon at sacredhearts.us (Shannon) Date: Thu, 25 Jun 2009 22:32:37 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> Message-ID: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> In case of bad battery in APC, are cdr's logged prior to system failure? On Thursday, June 25, 2009, Brian West wrote: > true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. > /b > On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: > Well this isn?t specific to FS crashing. The machine losing power would have the same effect, no??-Michael > Brian Westbrian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com? > > > > > -- Shannon From jingwei.yang at gmail.com Thu Jun 25 20:34:36 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 26 Jun 2009 11:34:36 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> Message-ID: <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> Hi Chris, here's the one that confuses me. As far as I understand, the extension 888 defined in public.xml is for picking up incoming calls. It should have no influence on outgoing calls, right? If not, what is to write to fit my case? (originate dingaling/gmail.com/userAAA at gmail.com&bridge(dingaling/ gmail.com/userBBB at gmail.com), both userAAA and userBBB can be internal or external). Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm not quite sure what to include. So I make it very simple. Here are three relative parameters in client.xml: Still, I got no echo for internal Ip calls. Please let me know where goes wrong. Thanks, -Jingwei On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen wrote: > I guess you have the problem here, > in client.xml you have > > > but you only define extension 888 in default context, > that's why nobody can reach you from public. > > under /usr/local/freeswitch/conf/dialplan > > define extension 888 in public.xml to the proper extension you expect, and > check the console log from fs_cli when you do gtalk calling to your gmail > client, you will find out the solution to your issue. > > chris > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/5fe6d7e1/attachment.html From brian at freeswitch.org Thu Jun 25 20:42:21 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:42:21 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> You could use something like nibble bill and at most loose the last interval of the call before its billed. You're going thru a lot of what if's ... You can't account for everything and you shouldn't have all your eggs in the same basket. /b On Jun 25, 2009, at 10:32 PM, Shannon wrote: > In case of bad battery in APC, are cdr's logged prior to system > failure? From mike at jerris.com Thu Jun 25 20:42:31 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 25 Jun 2009 23:42:31 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: <3BD9E49A-D794-4AB0-986D-082A54A39A3D@jerris.com> Of course not. This is why many do billing in icrements like mod_nibblebill does. Radius (although not yet with our module) and diamater both work this way and solve this issue. This in combination with session timers adress this and the hangup issue during a catastophic switch or network failure. On Jun 25, 2009, at 11:32 PM, Shannon wrote: > In case of bad battery in APC, are cdr's logged prior to system > failure? > > On Thursday, June 25, 2009, Brian West wrote: >> true dat... but again our goal is to not crash in the first >> place :P... nice APC can take care of the no power thing. >> /b >> On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: >> Well this isn?t specific to FS crashing. The machine losing power >> would have the same effect, no? -Michael >> Brian Westbrian at freeswitch.org >> -- Meet us at ClueCon! http://www.cluecon.com > > >> >> >> >> >> > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From harmeet at litatel.com Thu Jun 25 20:48:45 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Thu, 25 Jun 2009 23:48:45 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: Just make sure the power is always there! ...I know that some parts of the world this is not easy to achieve. Harmeet On Thu, Jun 25, 2009 at 11:32 PM, Shannon wrote: > In case of bad battery in APC, are cdr's logged prior to system failure? > > On Thursday, June 25, 2009, Brian West wrote: > > true dat... but again our goal is to not crash in the first place :P... > nice APC can take care of the no power thing. > > /b > > On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: > > Well this isn?t specific to FS crashing. The machine losing power would > have the same effect, no? -Michael > > Brian Westbrian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/f08effe1/attachment-0001.html From harmeet at litatel.com Thu Jun 25 20:53:13 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Thu, 25 Jun 2009 23:53:13 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> Message-ID: Does nibblebill update balances in real-time for each and every call? Does it do every second (micro/nano second?). How does it affect the performance vs if its done at end of call? I know that is not desirable for calling card applications. Harmeet On Thu, Jun 25, 2009 at 11:42 PM, Brian West wrote: > You could use something like nibble bill and at most loose the last > interval of the call before its billed. You're going thru a lot of > what if's ... You can't account for everything and you shouldn't have > all your eggs in the same basket. > > > > /b > > On Jun 25, 2009, at 10:32 PM, Shannon wrote: > > > In case of bad battery in APC, are cdr's logged prior to system > > failure? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/ffa73fa6/attachment.html From brian at freeswitch.org Thu Jun 25 20:56:07 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:56:07 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> Message-ID: <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> Well you would do it every 20-60 seconds maybe... It would be silly to do it every micro/nano second... it would cost you more in cpu and you don't gain much. /b On Jun 25, 2009, at 10:53 PM, Harmeet Singh wrote: > Does nibblebill update balances in real-time for each and every > call? Does it do every second (micro/nano second?). How does it > affect the performance vs if its done at end of call? I know that is > not desirable for calling card applications. > > Harmeet From brian at freeswitch.org Thu Jun 25 20:56:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:56:46 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> Is solar an option? ;) /b On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote: > Just make sure the power is always there! ...I know that some parts > of the world this is not easy to achieve. > > Harmeet From harmeet at litatel.com Thu Jun 25 21:06:26 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Fri, 26 Jun 2009 00:06:26 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> Message-ID: Can the interval be easily configures based on the destination? Like small interval for destinations with cost per minute > $1.00 and large intervals for cheaper destinations? Harmeet On Thu, Jun 25, 2009 at 11:56 PM, Brian West wrote: > Well you would do it every 20-60 seconds maybe... It would be silly to > do it every micro/nano second... it would cost you more in cpu and you > don't gain much. > > /b > > On Jun 25, 2009, at 10:53 PM, Harmeet Singh wrote: > > > Does nibblebill update balances in real-time for each and every > > call? Does it do every second (micro/nano second?). How does it > > affect the performance vs if its done at end of call? I know that is > > not desirable for calling card applications. > > > > Harmeet > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/a2798cb9/attachment.html From sprice at gmail.com Thu Jun 25 21:07:03 2009 From: sprice at gmail.com (SP) Date: Thu, 25 Jun 2009 23:07:03 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> Message-ID: <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> clouds On Thu, Jun 25, 2009 at 22:56, Brian West wrote: > Is solar an option? ;) > > /b > > On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote: > > > Just make sure the power is always there! ...I know that some parts > > of the world this is not easy to achieve. > > > > Harmeet > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/0cb8a59d/attachment.html From brian at freeswitch.org Thu Jun 25 21:11:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 23:11:15 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> Message-ID: <92D78B64-4AB7-442B-8596-DDA0D54AA6B9@freeswitch.org> yes you could. why not check it out and set it up ... its rather powerful. /b On Jun 25, 2009, at 11:06 PM, Harmeet Singh wrote: > Can the interval be easily configures based on the destination? Like > small interval for destinations with cost per minute > $1.00 and > large intervals for cheaper destinations? > > Harmeet From brian at freeswitch.org Thu Jun 25 21:11:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 23:11:31 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> Message-ID: <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> I also forgot about Nights. /b On Jun 25, 2009, at 11:07 PM, SP wrote: > clouds > > On Thu, Jun 25, 2009 at 22:56, Brian West > wrote: > Is solar an option? ;) > > /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/3f557c89/attachment.html From intralanman at freeswitch.org Thu Jun 25 21:51:34 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 26 Jun 2009 00:51:34 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> Message-ID: <4A4453D6.4060107@freeswitch.org> windmills From darklion11 at yahoo.com Thu Jun 25 22:48:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 22:48:01 -0700 (PDT) Subject: [Freeswitch-users] multiple gateways not working? Message-ID: <24215324.post@talk.nabble.com> --> --> Is this correct for multiple gateways? When I try this the first gateway works but the second gateway does not work? What is the solution for this can u help me? Thanks -- View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dome at tel.co.th Thu Jun 25 23:13:44 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:13:44 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215324.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> Message-ID: <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> May be need before first bridge Dome C. 2009/6/26 Edmar Cruz : > > > ? > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? ?--> > ? ? ? > ? ? > > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? ?--> > ? ? ? > ? ? > > ? > > > > Is this correct for multiple gateways? When I try this the first gateway > works but the second gateway does not work? > > > What is the solution for this can u help me? > > > Thanks > > -- > View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Thu Jun 25 23:31:40 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 26 Jun 2009 16:31:40 +1000 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> Message-ID: <20090626063140.GA13942@jdc.jasonjgw.net> Dome Charoenyost wrote: > May be need > > before first bridge and also, reading this wiki page may help http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate (see the discussion of multiple destinations) From darklion11 at yahoo.com Thu Jun 25 23:34:06 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 23:34:06 -0700 (PDT) Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> Message-ID: <24215631.post@talk.nabble.com> I try what you said still not working... Dome Charoenyost wrote: > > May be need > > before first bridge > > > Dome C. > 2009/6/26 Edmar Cruz : >> >> >> ? >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> ? ? ?--> >> ? ? ? >> ? ? >> >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> ? ? ?--> >> ? ? ? >> ? ? >> >> ? >> >> >> >> Is this correct for multiple gateways? When I try this the first gateway >> works but the second gateway does not work? >> >> >> What is the solution for this can u help me? >> >> >> Thanks >> >> -- >> View this message in context: >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dome at tel.co.th Thu Jun 25 23:45:02 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:45:02 +0700 Subject: [Freeswitch-users] From Asterisk to Freeswitch Message-ID: <8ccbff060906252345w1aa7c4a1u66f1c2af752dea20@mail.gmail.com> Dear All, I'm asterisk developer(I have some code in Asterisk) . After 3 weeks with freeswich nothing to say. now i'm move all callingcard , wholesale platfrom to Freeswitch. I'm very happy with bridge , nibblebill. after finish this job i'll test FS PBX feature. i think it's easy to do Hosted IPPBX. But i want to know more about mod_fifo Can someone tell me about mod_fifo compare with asterisk app_queue. i'm talking about annouce , priority agent Best Regards. Dome C. From dome at tel.co.th Thu Jun 25 23:46:14 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:46:14 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215631.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> Message-ID: <8ccbff060906252346r301469edg9350bc7091b12710@mail.gmail.com> Please try 2009/6/26 Edmar Cruz : > > > ? > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? > ? ? ? > ? ? > > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > > ? ? ? > ? ? > > ? > > > I try what you said still not working... > > > Dome Charoenyost wrote: >> >> May be need >> >> before first bridge >> >> >> Dome C. >> 2009/6/26 Edmar Cruz : >>> >>> >>> ? >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? >>> >>> >>> >>> Is this correct for multiple gateways? When I try this the first gateway >>> works but the second gateway does not work? >>> >>> >>> What is the solution for this can u help me? >>> >>> >>> Thanks >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dome at tel.co.th Thu Jun 25 23:48:03 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:48:03 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215631.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> Message-ID: <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> Or Try pipe if you want to ring all. Try comma 2009/6/26 Edmar Cruz : > > > ? > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? > ? ? ? > ? ? > > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > > ? ? ? > ? ? > > ? > > > I try what you said still not working... > > > Dome Charoenyost wrote: >> >> May be need >> >> before first bridge >> >> >> Dome C. >> 2009/6/26 Edmar Cruz : >>> >>> >>> ? >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? >>> >>> >>> >>> Is this correct for multiple gateways? When I try this the first gateway >>> works but the second gateway does not work? >>> >>> >>> What is the solution for this can u help me? >>> >>> >>> Thanks >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gcd at i.ph Thu Jun 25 23:49:26 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 26 Jun 2009 14:49:26 +0800 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215324.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> Message-ID: <7d0bfd8c0906252349ge0bf82dsa7f6c2250151f401@mail.gmail.com> you can combine the 2 gateways into one bridge app. pls see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall /nandy On Fri, Jun 26, 2009 at 1:48 PM, Edmar Cruz wrote: > > > > > > data="effective_caller_id_name=${effective_caller_id_name}"/> > data="effective_caller_id_number=${effective_caller_id_number}"/> > --> > > > > > > data="effective_caller_id_name=${effective_caller_id_name}"/> > data="effective_caller_id_number=${effective_caller_id_number}"/> > --> > > > > > > > > Is this correct for multiple gateways? When I try this the first gateway > works but the second gateway does not work? > > > What is the solution for this can u help me? > > > Thanks > > -- > View this message in context: > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/7b79db6e/attachment-0001.html From darklion11 at yahoo.com Fri Jun 26 00:05:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 26 Jun 2009 00:05:04 -0700 (PDT) Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> Message-ID: <24215893.post@talk.nabble.com> Yup your suggestions works... But I want my to have a prefix for the second bridge Dome Charoenyost wrote: > > Or Try pipe > > if you want to ring all. Try comma > > > > 2009/6/26 Edmar Cruz : >> >> >> ? >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> ? ? >> ? ? ? >> ? ? >> >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >> ? ? ? >> ? ? >> >> ? >> >> >> I try what you said still not working... >> >> >> Dome Charoenyost wrote: >>> >>> May be need >>> >>> before first bridge >>> >>> >>> Dome C. >>> 2009/6/26 Edmar Cruz : >>>> >>>> >>>> ? >>>> ? ? >>>> ? ? ? >>>> ? ? ?>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>>> ? ? ?>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>>> ? ? ?--> >>>> ? ? ?>>> data="sofia/default/$1 at 116.80.80.101"/> >>>> ? ? >>>> >>>> ? ? >>>> ? ? ? >>>> ? ? ?>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>>> ? ? ?>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>>> ? ? ?--> >>>> ? ? ?>>> data="sofia/default/$1 at 116.80.80.102"/> >>>> ? ? >>>> >>>> ? >>>> >>>> >>>> >>>> Is this correct for multiple gateways? When I try this the first >>>> gateway >>>> works but the second gateway does not work? >>>> >>>> >>>> What is the solution for this can u help me? >>>> >>>> >>>> Thanks >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215893.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Fri Jun 26 02:03:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 26 Jun 2009 02:03:31 -0700 (PDT) Subject: [Freeswitch-users] Service_not_implemented for mobile phones? Message-ID: <24217116.post@talk.nabble.com> Hi, I receive an error message service not implemented sometimes when calling a mobile phone number but sometimes it works. What maybe rhe cause of this error? I already installed zfone, perfectly connect to two freeswitch and the one issue I got today is these can you help me guys? Thanks -- View this message in context: http://www.nabble.com/Service_not_implemented-for-mobile-phones--tp24217116p24217116.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tayeb.meftah at gmail.com Fri Jun 26 02:25:04 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 26 Jun 2009 09:25:04 +0000 Subject: [Freeswitch-users] Freeswitch Calling cart application Message-ID: <4A4493F0.2040400@gmail.com> hello all, please i need a open source / free calling cart application to use with my freeswitch cool anyone chare any application with me? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4190 (20090626) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From mail at willboyce.com Fri Jun 26 03:17:41 2009 From: mail at willboyce.com (Will Boyce) Date: Fri, 26 Jun 2009 05:17:41 -0500 (CDT) Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <20344659.541246011373348.JavaMail.SYSTEM@man-00108> Message-ID: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> [ Optimised G.729A 'Howlet' for FreeSWITCH & Asterisk ] Howler Technologies are proud to announce today the launch of the first fully indemnified G.729A solution for FreeSWITCH. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source telecoms platforms. The G.729A Howlet ships as a drop-in module for FreeSWITCH or Asterisk, and enables cost-effective transcoding of G.729 and G.729A calls to other codecs. It scales to more than 500 concurrent transcoded calls on a single quad core server, and is licensed on a per concurrent channel basis. You can choose from two licensing models - fixed server perpetual and annual floating, the latter allowing you to 'float' your licensed channels across multiple servers for ultimate flexibility. Our unique floating licenses means you enable G.729A across your infrastructure without the administrative overhead of managing per-server licenses, and at a fraction of the initial cost of fixed-server licenses. Howlets are available for purchase immediately, and start at just ?3.99/channnel with all patent holder royalties taken care of. Download your free trial today ! http://www.howlertech.com/products/howlets/ -- Enjoy, The Howler Team < support at howlertech.com > Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7099 fax: +44 207 099 7098 http://www.howlertech.com/ Registered in England & Wales, Company No. 06285634 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/5638c104/attachment.html From jason at jasonjgw.net Fri Jun 26 03:33:49 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 26 Jun 2009 20:33:49 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090525011736.GA25198@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> Message-ID: <20090626103349.GA25435@jdc.jasonjgw.net> I can report that this problem (the failure of mod_portaudio.so to be linked properly) still persists as of revision 13970. The operating system is Debian Testing, and the difficulty began after upgrading from Libtool 1 to Libtool 2.2.6a. If anyone else can reproduce this or suggest a means of tracking down the cause, this would be much appreciated. The result of running ldd -r on mod_portaudio.so is as follows. It shows that most of the undefined symbols are from the Alsa library. I have also searched my build logs, (even with VERBOSE=1) but without locating any output that seems suspect. linux-vdso.so.1 => (0x00007fff279ff000) libm.so.6 => /lib/libm.so.6 (0x00007f261f415000) libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0x00007f261efd5000) libpthread.so.0 => /lib/libpthread.so.0 (0x00007f261edb9000) libc.so.6 => /lib/libc.so.6 (0x00007f261ea66000) libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f261e814000) libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f261e473000) libncurses.so.5 => /lib/libncurses.so.5 (0x00007f261e234000) libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f261df25000) libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f261dd09000) libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f261daa9000) /lib64/ld-linux-x86-64.so.2 (0x00007f261f8ee000) libdl.so.2 => /lib/libdl.so.2 (0x00007f261d8a5000) libz.so.1 => /usr/lib/libz.so.1 (0x00007f261d68d000) libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f261d484000) undefined symbol: crypt_r (/opt/freeswitch/lib/libfreeswitch.so.1) undefined symbol: clock_gettime (/opt/freeswitch/lib/libfreeswitch.so.1) undefined symbol: snd_config (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_period_size_min (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_channels (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_period_size_near (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_rate_near (./mod_portaudio.so) undefined symbol: clock_gettime (./mod_portaudio.so) undefined symbol: snd_pcm_poll_descriptors_revents (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_access (./mod_portaudio.so) undefined symbol: snd_pcm_format_size (./mod_portaudio.so) undefined symbol: snd_strerror (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_periods_min (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_rate (./mod_portaudio.so) undefined symbol: snd_pcm_info_set_device (./mod_portaudio.so) undefined symbol: snd_pcm_status (./mod_portaudio.so) undefined symbol: snd_pcm_close (./mod_portaudio.so) undefined symbol: snd_lib_error_set_handler (./mod_portaudio.so) undefined symbol: snd_config_get_id (./mod_portaudio.so) undefined symbol: snd_pcm_avail_update (./mod_portaudio.so) undefined symbol: snd_pcm_info_set_subdevice (./mod_portaudio.so) undefined symbol: snd_pcm_area_copy (./mod_portaudio.so) undefined symbol: snd_pcm_state (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_channels_min (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_period_size (./mod_portaudio.so) undefined symbol: snd_pcm_info_get_card (./mod_portaudio.so) undefined symbol: snd_pcm_poll_descriptors_count (./mod_portaudio.so) undefined symbol: snd_ctl_open (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_buffer_size_min (./mod_portaudio.so) undefined symbol: snd_config_update (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_channels_max (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_test_format (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_buffer_size_max (./mod_portaudio.so) undefined symbol: snd_ctl_card_info_get_name (./mod_portaudio.so) undefined symbol: snd_pcm_status_sizeof (./mod_portaudio.so) undefined symbol: snd_pcm_delay (./mod_portaudio.so) undefined symbol: snd_pcm_drain (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_tstamp (./mod_portaudio.so) undefined symbol: snd_pcm_start (./mod_portaudio.so) undefined symbol: snd_pcm_open (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_periods_integer (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_test_access (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_silence_threshold (./mod_portaudio.so) undefined symbol: snd_pcm_areas_silence (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_format (./mod_portaudio.so) undefined symbol: snd_ctl_close (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_xfer_align (./mod_portaudio.so) undefined symbol: snd_pcm_mmap_commit (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_sizeof (./mod_portaudio.so) undefined symbol: snd_config_update_free_global (./mod_portaudio.so) undefined symbol: snd_pcm_mmap_begin (./mod_portaudio.so) undefined symbol: snd_ctl_card_info (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_get_boundary (./mod_portaudio.so) undefined symbol: snd_ctl_pcm_next_device (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_periods_max (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params (./mod_portaudio.so) undefined symbol: snd_config_get_string (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_any (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_state (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_test_period_size (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_current (./mod_portaudio.so) undefined symbol: snd_pcm_link (./mod_portaudio.so) undefined symbol: snd_config_iterator_entry (./mod_portaudio.so) undefined symbol: snd_pcm_poll_descriptors (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_buffer_size_near (./mod_portaudio.so) undefined symbol: snd_pcm_info_sizeof (./mod_portaudio.so) undefined symbol: snd_ctl_pcm_info (./mod_portaudio.so) undefined symbol: snd_pcm_nonblock (./mod_portaudio.so) undefined symbol: snd_config_search (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_rate_numden (./mod_portaudio.so) undefined symbol: snd_ctl_card_info_sizeof (./mod_portaudio.so) undefined symbol: snd_pcm_drop (./mod_portaudio.so) undefined symbol: snd_config_iterator_end (./mod_portaudio.so) undefined symbol: snd_config_iterator_next (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_start_threshold (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_trigger_tstamp (./mod_portaudio.so) undefined symbol: snd_pcm_info_set_stream (./mod_portaudio.so) undefined symbol: snd_pcm_info (./mod_portaudio.so) undefined symbol: snd_config_iterator_first (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_stop_threshold (./mod_portaudio.so) undefined symbol: snd_card_next (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_silence_size (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_delay (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_buffer_size (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_avail_min (./mod_portaudio.so) undefined symbol: snd_pcm_info_get_name (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params (./mod_portaudio.so) undefined symbol: snd_pcm_prepare (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_sizeof (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_period_size_max (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_tstamp_mode (./mod_portaudio.so) From jalsot at gmail.com Fri Jun 26 04:03:36 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 26 Jun 2009 13:03:36 +0200 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090626103349.GA25435@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> Message-ID: <4A44AB08.1050503@gmail.com> Did you make bootstrap.sh and configure before compilation? We have no issues on Ubuntu 9.04 (libtool 2.2.6a). Regards, Tamas Jason White ?rta: > I can report that this problem (the failure of mod_portaudio.so to be linked > properly) still persists as of revision 13970. > > The operating system is Debian Testing, and the difficulty began after > upgrading from Libtool 1 to Libtool 2.2.6a. > > If anyone else can reproduce this or suggest a means of tracking down the > cause, this would be much appreciated. > > The result of running ldd -r on mod_portaudio.so is as follows. It shows that > most of the undefined symbols are from the Alsa library. > > I have also searched my build logs, (even with VERBOSE=1) but without locating > any output that seems suspect. > > linux-vdso.so.1 => (0x00007fff279ff000) > libm.so.6 => /lib/libm.so.6 (0x00007f261f415000) > libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0x00007f261efd5000) > libpthread.so.0 => /lib/libpthread.so.0 (0x00007f261edb9000) > libc.so.6 => /lib/libc.so.6 (0x00007f261ea66000) > libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f261e814000) > libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f261e473000) > libncurses.so.5 => /lib/libncurses.so.5 (0x00007f261e234000) > libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f261df25000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f261dd09000) > libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f261daa9000) > /lib64/ld-linux-x86-64.so.2 (0x00007f261f8ee000) > libdl.so.2 => /lib/libdl.so.2 (0x00007f261d8a5000) > libz.so.1 => /usr/lib/libz.so.1 (0x00007f261d68d000) > libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f261d484000) > undefined symbol: crypt_r (/opt/freeswitch/lib/libfreeswitch.so.1) > undefined symbol: clock_gettime (/opt/freeswitch/lib/libfreeswitch.so.1) > undefined symbol: snd_config (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_period_size_min (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_channels (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_period_size_near (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_rate_near (./mod_portaudio.so) > undefined symbol: clock_gettime (./mod_portaudio.so) > undefined symbol: snd_pcm_poll_descriptors_revents (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_access (./mod_portaudio.so) > undefined symbol: snd_pcm_format_size (./mod_portaudio.so) > undefined symbol: snd_strerror (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_periods_min (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_rate (./mod_portaudio.so) > undefined symbol: snd_pcm_info_set_device (./mod_portaudio.so) > undefined symbol: snd_pcm_status (./mod_portaudio.so) > undefined symbol: snd_pcm_close (./mod_portaudio.so) > undefined symbol: snd_lib_error_set_handler (./mod_portaudio.so) > undefined symbol: snd_config_get_id (./mod_portaudio.so) > undefined symbol: snd_pcm_avail_update (./mod_portaudio.so) > undefined symbol: snd_pcm_info_set_subdevice (./mod_portaudio.so) > undefined symbol: snd_pcm_area_copy (./mod_portaudio.so) > undefined symbol: snd_pcm_state (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_channels_min (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_period_size (./mod_portaudio.so) > undefined symbol: snd_pcm_info_get_card (./mod_portaudio.so) > undefined symbol: snd_pcm_poll_descriptors_count (./mod_portaudio.so) > undefined symbol: snd_ctl_open (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_buffer_size_min (./mod_portaudio.so) > undefined symbol: snd_config_update (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_channels_max (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_test_format (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_buffer_size_max (./mod_portaudio.so) > undefined symbol: snd_ctl_card_info_get_name (./mod_portaudio.so) > undefined symbol: snd_pcm_status_sizeof (./mod_portaudio.so) > undefined symbol: snd_pcm_delay (./mod_portaudio.so) > undefined symbol: snd_pcm_drain (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_tstamp (./mod_portaudio.so) > undefined symbol: snd_pcm_start (./mod_portaudio.so) > undefined symbol: snd_pcm_open (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_periods_integer (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_test_access (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_silence_threshold (./mod_portaudio.so) > undefined symbol: snd_pcm_areas_silence (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_format (./mod_portaudio.so) > undefined symbol: snd_ctl_close (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_xfer_align (./mod_portaudio.so) > undefined symbol: snd_pcm_mmap_commit (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_sizeof (./mod_portaudio.so) > undefined symbol: snd_config_update_free_global (./mod_portaudio.so) > undefined symbol: snd_pcm_mmap_begin (./mod_portaudio.so) > undefined symbol: snd_ctl_card_info (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_get_boundary (./mod_portaudio.so) > undefined symbol: snd_ctl_pcm_next_device (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_periods_max (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params (./mod_portaudio.so) > undefined symbol: snd_config_get_string (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_any (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_state (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_test_period_size (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_current (./mod_portaudio.so) > undefined symbol: snd_pcm_link (./mod_portaudio.so) > undefined symbol: snd_config_iterator_entry (./mod_portaudio.so) > undefined symbol: snd_pcm_poll_descriptors (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_buffer_size_near (./mod_portaudio.so) > undefined symbol: snd_pcm_info_sizeof (./mod_portaudio.so) > undefined symbol: snd_ctl_pcm_info (./mod_portaudio.so) > undefined symbol: snd_pcm_nonblock (./mod_portaudio.so) > undefined symbol: snd_config_search (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_rate_numden (./mod_portaudio.so) > undefined symbol: snd_ctl_card_info_sizeof (./mod_portaudio.so) > undefined symbol: snd_pcm_drop (./mod_portaudio.so) > undefined symbol: snd_config_iterator_end (./mod_portaudio.so) > undefined symbol: snd_config_iterator_next (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_start_threshold (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_trigger_tstamp (./mod_portaudio.so) > undefined symbol: snd_pcm_info_set_stream (./mod_portaudio.so) > undefined symbol: snd_pcm_info (./mod_portaudio.so) > undefined symbol: snd_config_iterator_first (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_stop_threshold (./mod_portaudio.so) > undefined symbol: snd_card_next (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_silence_size (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_delay (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_buffer_size (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_avail_min (./mod_portaudio.so) > undefined symbol: snd_pcm_info_get_name (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params (./mod_portaudio.so) > undefined symbol: snd_pcm_prepare (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_sizeof (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_period_size_max (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_tstamp_mode (./mod_portaudio.so) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From odermann at googlemail.com Fri Jun 26 04:38:12 2009 From: odermann at googlemail.com (Dennis) Date: Fri, 26 Jun 2009 13:38:12 +0200 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud Message-ID: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> hi, we want to use uuid_displace with mux to playback a soundfile to a bridged uuid, so that this uuid can hear the other side talk AND hear the soundfile (whispering). is there an option we can set, for defining the loudness level of the soundfile? in our tests the soundfile was way to loud, so that it was nearly impossible to hear the other side talk, while the soundfile was playing. we tried "uuid_displace uuid start /path/to/soundfile/soundfile.wav 0 mux 0.3", so that the loudness of soundfile only would be 30% - but this does not work. thanks & kind regards dennis From harmeet at litatel.com Fri Jun 26 05:10:15 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Fri, 26 Jun 2009 08:10:15 -0400 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215893.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> <24215893.post@talk.nabble.com> Message-ID: Just add the prefix like - BTW: Never give your real prefix. Anybody can use it to send traffic to your trunk with that prefix and eat away your balance. On Fri, Jun 26, 2009 at 3:05 AM, Edmar Cruz wrote: > > Yup your suggestions works... But I want my to have a prefix for the second > bridge > > > > > > Dome Charoenyost wrote: > > > > Or Try pipe > > > > if you want to ring all. Try comma > > > > > > > > 2009/6/26 Edmar Cruz : > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_name=${effective_caller_id_name}"/> > >> >> data="effective_caller_id_number=${effective_caller_id_number}"/> > >> > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_name=${effective_caller_id_name}"/> > >> >> data="effective_caller_id_number=${effective_caller_id_number}"/> > >> > >> > >> > >> > >> > >> > >> > >> I try what you said still not working... > >> > >> > >> Dome Charoenyost wrote: > >>> > >>> May be need > >>> > >>> before first bridge > >>> > >>> > >>> Dome C. > >>> 2009/6/26 Edmar Cruz : > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> > >>>> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> > >>>> --> > >>>> >>>> data="sofia/default/$1 at 116.80.80.101"/> > >>>> > >>>> > >>>> > >>>> > >>>> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> > >>>> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> > >>>> --> > >>>> >>>> data="sofia/default/$1 at 116.80.80.102"/> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> Is this correct for multiple gateways? When I try this the first > >>>> gateway > >>>> works but the second gateway does not work? > >>>> > >>>> > >>>> What is the solution for this can u help me? > >>>> > >>>> > >>>> Thanks > >>>> > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215893.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/e4a30246/attachment.html From harmeet at litatel.com Fri Jun 26 05:23:04 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Fri, 26 Jun 2009 08:23:04 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <4A4453D6.4060107@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> <4A4453D6.4060107@freeswitch.org> Message-ID: Pedals! For countries with cheap labour, this can give endless power! Each pedal connected to a dynamo. Workers come in shift after shift and keep pedalling! Its a solution for unemployement and electricity in those countries! Check this out - http://www.msnbc.msn.com/id/26430304 On Fri, Jun 26, 2009 at 12:51 AM, Raymond Chandler < intralanman at freeswitch.org> wrote: > windmills > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/744c2c73/attachment.html From excelsio at gmx.net Fri Jun 26 05:33:37 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Fri, 26 Jun 2009 14:33:37 +0200 Subject: [Freeswitch-users] [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. Message-ID: <20090626123337.243300@gmx.net> [eap] EAP NAK [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. Hi, having a new voip pbx (OmniPCX Enterprise 9.0) from Alcatel-Lucent, I now try to setup 802.1x with the phones, an Alcatel-Lucent IP Touch 4028 EE. Freeradius is 2.x on a Debian 5.0. My first attempt was with MD5, which works without any problem. Next step is TLS, which works at 50%. Well, the client authentication of TLS works, but when I configure to do a server authentication within the IP phone?s setup, it fails. So, here is the output from radiusd -Xf when server authentication is not added within the IP phone?s setup: ============================================================= FreeRADIUS Version 2.1.6, for host i686-pc-linux-gnu, built on May 29 2009 at 15:54:08 Copyright (C) 1999-2009 The FreeRADIUS server project and contributors. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. You may redistribute copies of FreeRADIUS under the terms of the GNU General Public License v2. Starting - reading configuration files ... including configuration file /etc/raddb/radiusd.conf including configuration file /etc/raddb/proxy.conf including configuration file /etc/raddb/clients.conf including files in directory /etc/raddb/modules/ including configuration file /etc/raddb/modules/radutmp including configuration file /etc/raddb/modules/pap including configuration file /etc/raddb/modules/attr_rewrite including configuration file /etc/raddb/modules/perl including configuration file /etc/raddb/modules/expr including configuration file /etc/raddb/modules/smbpasswd including configuration file /etc/raddb/modules/otp including configuration file /etc/raddb/modules/acct_unique including configuration file /etc/raddb/modules/chap including configuration file /etc/raddb/modules/krb5 including configuration file /etc/raddb/modules/files including configuration file /etc/raddb/modules/policy including configuration file /etc/raddb/modules/wimax including configuration file /etc/raddb/modules/ldap including configuration file /etc/raddb/modules/realm including configuration file /etc/raddb/modules/mschap including configuration file /etc/raddb/modules/linelog including configuration file /etc/raddb/modules/sql_log including configuration file /etc/raddb/modules/detail including configuration file /etc/raddb/modules/digest including configuration file /etc/raddb/modules/mac2vlan including configuration file /etc/raddb/modules/expiration including configuration file /etc/raddb/modules/echo including configuration file /etc/raddb/modules/detail.example.com including configuration file /etc/raddb/modules/counter including configuration file /etc/raddb/modules/attr_filter including configuration file /etc/raddb/modules/sradutmp including configuration file /etc/raddb/modules/pam including configuration file /etc/raddb/modules/inner-eap including configuration file /etc/raddb/modules/smsotp including configuration file /etc/raddb/modules/passwd including configuration file /etc/raddb/modules/ippool including configuration file /etc/raddb/modules/logintime including configuration file /etc/raddb/modules/always including configuration file /etc/raddb/modules/preprocess including configuration file /etc/raddb/modules/sqlcounter_expire_on_login including configuration file /etc/raddb/modules/exec including configuration file /etc/raddb/modules/unix including configuration file /etc/raddb/modules/etc_group including configuration file /etc/raddb/modules/mac2ip including configuration file /etc/raddb/modules/checkval including configuration file /etc/raddb/modules/detail.log including configuration file /etc/raddb/eap.conf including configuration file /etc/raddb/policy.conf including files in directory /etc/raddb/sites-enabled/ including configuration file /etc/raddb/sites-enabled/inner-tunnel including configuration file /etc/raddb/sites-enabled/default including configuration file /etc/raddb/sites-enabled/control-socket including dictionary file /etc/raddb/dictionary main { prefix = "/usr" localstatedir = "/var" logdir = "/var/log/radius" libdir = "/usr/lib/freeradius" radacctdir = "/var/log/radius/radacct" hostname_lookups = no max_request_time = 30 cleanup_delay = 5 max_requests = 1024 allow_core_dumps = no pidfile = "/var/run/radiusd/radiusd.pid" checkrad = "/usr/sbin/checkrad" debug_level = 0 proxy_requests = yes log { stripped_names = no auth = no auth_badpass = no auth_goodpass = no } security { max_attributes = 200 reject_delay = 1 status_server = yes } } radiusd: #### Loading Realms and Home Servers #### proxy server { retry_delay = 5 retry_count = 3 default_fallback = no dead_time = 120 wake_all_if_all_dead = no } home_server localhost { ipaddr = 127.0.0.1 port = 1812 type = "auth" secret = "testing123" response_window = 20 max_outstanding = 65536 require_message_authenticator = no zombie_period = 40 status_check = "status-server" ping_interval = 30 check_interval = 30 num_answers_to_alive = 3 num_pings_to_alive = 3 revive_interval = 120 status_check_timeout = 4 irt = 2 mrt = 16 mrc = 5 mrd = 30 } home_server_pool my_auth_failover { type = fail-over home_server = localhost } realm example.com { auth_pool = my_auth_failover } realm LOCAL { } radiusd: #### Loading Clients #### client localhost { ipaddr = 127.0.0.1 require_message_authenticator = no secret = "testing123" nastype = "other" } client 192.168.10.130 { require_message_authenticator = no secret = "password" shortname = "Switch1" } client 192.168.10.131 { require_message_authenticator = no secret = "password" shortname = "Switch2" } client 192.168.104.244 { require_message_authenticator = no secret = "password" shortname = "Switch3" } radiusd: #### Instantiating modules #### instantiate { Module: Linked to module rlm_exec Module: Instantiating exec exec { wait = no input_pairs = "request" shell_escape = yes } Module: Linked to module rlm_expr Module: Instantiating expr Module: Linked to module rlm_expiration Module: Instantiating expiration expiration { reply-message = "Password Has Expired " } Module: Linked to module rlm_logintime Module: Instantiating logintime logintime { reply-message = "You are calling outside your allowed timespan " minimum-timeout = 60 } } radiusd: #### Loading Virtual Servers #### server inner-tunnel { modules { Module: Checking authenticate {...} for more modules to load Module: Linked to module rlm_pap Module: Instantiating pap pap { encryption_scheme = "auto" auto_header = no } Module: Linked to module rlm_chap Module: Instantiating chap Module: Linked to module rlm_mschap Module: Instantiating mschap mschap { use_mppe = yes require_encryption = no require_strong = no with_ntdomain_hack = no } Module: Linked to module rlm_unix Module: Instantiating unix unix { radwtmp = "/var/log/radius/radwtmp" } Module: Linked to module rlm_eap Module: Instantiating eap eap { default_eap_type = "tls" timer_expire = 60 ignore_unknown_eap_types = no cisco_accounting_username_bug = no max_sessions = 2048 } Module: Linked to sub-module rlm_eap_md5 Module: Instantiating eap-md5 Module: Linked to sub-module rlm_eap_leap Module: Instantiating eap-leap Module: Linked to sub-module rlm_eap_gtc Module: Instantiating eap-gtc gtc { challenge = "Password: " auth_type = "PAP" } Module: Linked to sub-module rlm_eap_tls Module: Instantiating eap-tls tls { rsa_key_exchange = yes dh_key_exchange = yes rsa_key_length = 2048 dh_key_length = 512 verify_depth = 0 pem_file_type = yes private_key_file = "/etc/raddb/certs/server.key" certificate_file = "/etc/raddb/certs/server.crt" CA_file = "/etc/raddb/certs/ca.crt" private_key_password = "password" dh_file = "/etc/raddb/certs/dh" random_file = "/etc/raddb/certs/random" fragment_size = 1024 include_length = yes check_crl = no cipher_list = "DEFAULT" make_cert_command = "/etc/raddb/certs/bootstrap" cache { enable = no lifetime = 24 max_entries = 255 } } Module: Linked to sub-module rlm_eap_ttls Module: Instantiating eap-ttls ttls { default_eap_type = "md5" copy_request_to_tunnel = yes use_tunneled_reply = yes virtual_server = "inner-tunnel" include_length = yes } Module: Linked to sub-module rlm_eap_peap Module: Instantiating eap-peap peap { default_eap_type = "mschapv2" copy_request_to_tunnel = no use_tunneled_reply = no proxy_tunneled_request_as_eap = yes virtual_server = "inner-tunnel" } Module: Linked to sub-module rlm_eap_mschapv2 Module: Instantiating eap-mschapv2 mschapv2 { with_ntdomain_hack = no } Module: Checking authorize {...} for more modules to load Module: Linked to module rlm_realm Module: Instantiating suffix realm suffix { format = "suffix" delimiter = "@" ignore_default = no ignore_null = no } Module: Linked to module rlm_files Module: Instantiating files files { usersfile = "/etc/raddb/users" acctusersfile = "/etc/raddb/acct_users" preproxy_usersfile = "/etc/raddb/preproxy_users" compat = "no" } Module: Checking session {...} for more modules to load Module: Linked to module rlm_radutmp Module: Instantiating radutmp radutmp { filename = "/var/log/radius/radutmp" username = "%{User-Name}" case_sensitive = yes check_with_nas = yes perm = 384 callerid = yes } Module: Checking post-proxy {...} for more modules to load Module: Checking post-auth {...} for more modules to load Module: Linked to module rlm_attr_filter Module: Instantiating attr_filter.access_reject attr_filter attr_filter.access_reject { attrsfile = "/etc/raddb/attrs.access_reject" key = "%{User-Name}" } } # modules } # server server { modules { Module: Checking authenticate {...} for more modules to load Module: Checking authorize {...} for more modules to load Module: Linked to module rlm_preprocess Module: Instantiating preprocess preprocess { huntgroups = "/etc/raddb/huntgroups" hints = "/etc/raddb/hints" with_ascend_hack = no ascend_channels_per_line = 23 with_ntdomain_hack = no with_specialix_jetstream_hack = no with_cisco_vsa_hack = no with_alvarion_vsa_hack = no } Module: Checking preacct {...} for more modules to load Module: Linked to module rlm_acct_unique Module: Instantiating acct_unique acct_unique { key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port" } Module: Checking accounting {...} for more modules to load Module: Linked to module rlm_detail Module: Instantiating detail detail { detailfile = "/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d" header = "%t" detailperm = 384 dirperm = 493 locking = no log_packet_header = no } Module: Instantiating attr_filter.accounting_response attr_filter attr_filter.accounting_response { attrsfile = "/etc/raddb/attrs.accounting_response" key = "%{User-Name}" } Module: Checking session {...} for more modules to load Module: Checking post-proxy {...} for more modules to load Module: Checking post-auth {...} for more modules to load } # modules } # server radiusd: #### Opening IP addresses and Ports #### listen { type = "auth" ipaddr = * port = 0 } listen { type = "acct" ipaddr = * port = 0 } listen { type = "control" listen { socket = "/var/run/radiusd/radiusd.sock" } } Listening on authentication address * port 1812 Listening on accounting address * port 1813 Listening on command file /var/run/radiusd/radiusd.sock Listening on proxy address * port 1814 Ready to process requests. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=127, length=336 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" EAP-Message = 0x0244001101303038303966383538633137 Message-Authenticator = 0x26f23798cfc04abb83fa87d210d1bc69 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 68 length 17 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop [pap] WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] EAP Identity [eap] processing type tls [tls] Requiring client certificate [tls] Initiate [tls] Start returned 1 ++[eap] returns handled Sending Access-Challenge of id 127 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x014500060d20 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aa8262fa7eff5a1519b312199 Finished request 0. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=128, length=437 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aa8262fa7eff5a1519b312199 EAP-Message = 0x024500640d800000005a160301005501000051030100000000029c3cefd84c43ff0fdd96f2139986e55316e45f34fe5e36a3caa07f00002a00660065006400630062006100600015001400120011000900080006000500040003001a0019001800170100 Message-Authenticator = 0xca28b57e66136938c479c4f7c87ca2b9 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 69 length 100 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS TLS Length 90 [tls] Length Included [tls] eaptls_verify returned 11 [tls] (other): before/accept initialization [tls] TLS_accept: before/accept initialization [tls] <<< TLS 1.0 Handshake [length 0055], ClientHello [tls] TLS_accept: SSLv3 read client hello A [tls] >>> TLS 1.0 Handshake [length 002a], ServerHello [tls] TLS_accept: SSLv3 write server hello A [tls] >>> TLS 1.0 Handshake [length 03e2], Certificate [tls] TLS_accept: SSLv3 write certificate A [tls] >>> TLS 1.0 Handshake [length 00c2], CertificateRequest [tls] TLS_accept: SSLv3 write certificate request A [tls] TLS_accept: SSLv3 flush data [tls] TLS_accept: Need to read more data: SSLv3 read client certificate A In SSL Handshake Phase In SSL Accept mode [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 128 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 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 EAP-Message = 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 EAP-Message = 0x82010f003082010a0282010100bdfe6eadd2eb98a40254308cecc6ae20cf243c0c8006842b54e3f584836d32feb01a3defad2df56e44f3e914eed1638e9743aad8ce964cf59dea4a21ea2643a2c2fe689c4e4f0ec30440a53f21039d6f53fa1f1ccbd18de6cf6d54041b21335042ede6508fb50983159e8bc8392936682c76053230369d77d8f7021b4851f3220263819145f78d98593bd488d874ecb5ba2179a888a746d8725c0aa910412904f876716322c44d90fc02e1aded3c6b3be48920b0998e5910bc835b75d592d6f3ccc2866952bf3d1c40b4799a527b54353338fb7cd7cdd0acbd42a1c7904a5a32d366704e95923c9d826ed49de49160e7 EAP-Message = 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 EAP-Message = 0x710afb0e1495d38c42da7dd3 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aa9252fa7eff5a1519b312199 Finished request 1. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=129, length=343 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aa9252fa7eff5a1519b312199 EAP-Message = 0x024600060d00 Message-Authenticator = 0x5a31886be829f7fe67e6f24f7d17cd3f MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 70 length 6 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS [tls] Received TLS ACK [tls] ACK handshake fragment handler [tls] eaptls_verify returned 1 [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 129 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x014700f10d80000004dd8a8ab4bce56d175b1c0969bdd410271ff5999d0d1d3fb011bdfbe4504764c0b116030100c20d0000ba0301024000b400b23081af310b30090603550406130244453110300e0603550407130742616d62657267312c302a060355040a13234f74746f2d4672696564726963682d556e697665727369746165742042616d6265726731163014060355040b130d52656368656e7a656e7472756d311c301a060355040313134b6f6d6d756e696b6174696f6e736e65747a65312a302806092a864886f70d010901161b6e65747a2d7365727669636540756e692d62616d626572672e64650e000000 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aaa242fa7eff5a1519b312199 Finished request 2. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=130, length=1657 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aaa242fa7eff5a1519b312199 EAP-Message = 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 EAP-Message = 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 EAP-Message = 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 EAP-Message = 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 EAP-Message = 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 EAP-Message = 0xafb637c54c0f2af5ce0b900abaa00583a06ef6f041d6bdd639497450a2f527e4a0b53e41ee93ef1311eab231c6 Message-Authenticator = 0xa32aab76eea9d3033e8ea581033ddf23 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 71 length 253 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS TLS Length 1560 [tls] Received EAP-TLS First Fragment of the message [tls] eaptls_verify returned 9 [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 130 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x014800060d00 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aab2b2fa7eff5a1519b312199 Finished request 3. Going to the next request Waking up in 3.7 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=131, length=605 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aab2b2fa7eff5a1519b312199 EAP-Message = 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 EAP-Message = 0x14e12d8625cc5ce37e5e548243 Message-Authenticator = 0x5d76a81c99d9ed8d117658f6693d21f8 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 72 length 253 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS [tls] eaptls_verify returned 7 [tls] Done initial handshake [tls] <<< TLS 1.0 Handshake [length 03ca], Certificate [tls] chain-depth=1, [tls] error=0 [tls] --> User-Name = 00809f858c17 [tls] --> BUF-Name = [tls] --> subject = /C=DE/ [tls] --> issuer = /C=DE/ [tls] --> verify return:1 [tls] chain-depth=0, [tls] error=0 [tls] --> User-Name = 00809f858c17 [tls] --> BUF-Name = 3850 [tls] --> subject = /C=DE/L=[tls] --> issuer = /C=DE/L= [tls] --> verify return:1 [tls] TLS_accept: SSLv3 read client certificate A [tls] <<< TLS 1.0 Handshake [length 0106], ClientKeyExchange [tls] TLS_accept: SSLv3 read client key exchange A [tls] <<< TLS 1.0 Handshake [length 0106], CertificateVerify [tls] TLS_accept: SSLv3 read certificate verify A [tls] <<< TLS 1.0 ChangeCipherSpec [length 0001] [tls] <<< TLS 1.0 Handshake [length 0010], Finished [tls] TLS_accept: SSLv3 read finished A [tls] >>> TLS 1.0 ChangeCipherSpec [length 0001] [tls] TLS_accept: SSLv3 write change cipher spec A [tls] >>> TLS 1.0 Handshake [length 0010], Finished [tls] TLS_accept: SSLv3 write finished A [tls] TLS_accept: SSLv3 flush data [tls] (other): SSL negotiation finished successfully SSL Connection Established [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 131 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x0149003d0d800000003314030100010116030100284751a33290290d3f84335c77caabe21228593b3c56db90e09d9bb1e672ef3a5285b5e7638d2931e5 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aac2a2fa7eff5a1519b312199 Finished request 4. Going to the next request Waking up in 3.6 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=132, length=343 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aac2a2fa7eff5a1519b312199 EAP-Message = 0x024900060d00 Message-Authenticator = 0xd9f31890976547a84f346a88f7c1d87f MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 73 length 6 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS [tls] Received TLS ACK [tls] ACK handshake is finished [tls] eaptls_verify returned 3 [tls] eaptls_process returned 3 [tls] Adding user data to cached session [eap] Freeing handler ++[eap] returns ok +- entering group post-auth {...} ++[exec] returns noop Sending Access-Accept of id 132 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP MS-MPPE-Recv-Key = 0x78b733f877a98c69f4197d25508c9528eb2cd0d68686ce27dffeb875d0550248 MS-MPPE-Send-Key = 0x5893ecdda559f6e70bafffb3088f4329608a138b74c62db3bf4eba8a187daf2d EAP-Message = 0x03490004 Message-Authenticator = 0x00000000000000000000000000000000 User-Name = "00809f858c17" Finished request 5. Going to the next request Waking up in 3.6 seconds. Cleaning up request 0 ID 127 with timestamp +2 Cleaning up request 1 ID 128 with timestamp +2 Cleaning up request 2 ID 129 with timestamp +2 Waking up in 1.2 seconds. Cleaning up request 3 ID 130 with timestamp +4 Cleaning up request 4 ID 131 with timestamp +4 Cleaning up request 5 ID 132 with timestamp +4 Ready to process requests. ============================================================================================================================= As soon as I enable "Server Authentication" wthin the IP phone, it fails: ============================================================================================================================= Going to the next request Ready to process requests. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=146, length=336 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" EAP-Message = 0x0217001101303038303966383538633137 Message-Authenticator = 0xb2b310ea9fb4000f3c2a436a6646b653 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 23 length 17 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop [pap] WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] EAP Identity [eap] processing type tls [tls] Requiring client certificate [tls] Initiate [tls] Start returned 1 ++[eap] returns handled Sending Access-Challenge of id 146 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x011800060d20 Message-Authenticator = 0x00000000000000000000000000000000 State = 0x750d139875151e34b18bc5eb9140ffe1 Finished request 9. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=147, length=343 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0x750d139875151e34b18bc5eb9140ffe1 EAP-Message = 0x02180006030d Message-Authenticator = 0xc861eecfeba7db7c00683c189e63b265 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 24 length 6 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop [pap] WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP NAK [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. [eap] No common EAP types found. [eap] Failed in EAP select ++[eap] returns invalid Failed to authenticate the user. Using Post-Auth-Type Reject +- entering group REJECT {...} [attr_filter.access_reject] expand: %{User-Name} -> 00809f858c17 attr_filter: Matched entry DEFAULT at line 11 ++[attr_filter.access_reject] returns updated Delaying reject of request 10 for 1 seconds Going to the next request Waking up in 0.9 seconds. Sending delayed reject for request 10 Sending Access-Reject of id 147 to 192.168.10.130 port 1812 EAP-Message = 0x04180004 Message-Authenticator = 0x00000000000000000000000000000000 Waking up in 3.9 seconds. Cleaning up request 9 ID 146 with timestamp +131 Waking up in 1.0 seconds. Cleaning up request 10 ID 147 with timestamp +131 Ready to process requests. Well, what?s going wrong? Michael From excelsio at gmx.net Fri Jun 26 05:38:33 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Fri, 26 Jun 2009 14:38:33 +0200 Subject: [Freeswitch-users] [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. In-Reply-To: <20090626123337.243300@gmx.net> References: <20090626123337.243300@gmx.net> Message-ID: <20090626123833.103370@gmx.net> Sorry, wrong group :-) Freeswitch, Freeradius, both free sorry :-) From woof at iwoof.org Fri Jun 26 06:58:53 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 26 Jun 2009 09:58:53 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <4A4453D6.4060107@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> <4A4453D6.4060107@freeswitch.org> Message-ID: On Fri, 26 Jun 2009 00:51:34 -0400, Raymond Chandler wrote: > windmills Just set them up to be right next to the output of the system's fan! I gotta get to the patent office, quick! --Woof! From intralanman at freeswitch.org Fri Jun 26 07:12:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 26 Jun 2009 10:12:15 -0400 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> Message-ID: <4A44D73F.2010408@freeswitch.org> There doesn't seem to be any direct link between FreeSWITCH and Howler. How does this benefit the project? Can we assume that Howler is "giving something back"? -Ray From dave at 3c.co.uk Fri Jun 26 07:38:43 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 26 Jun 2009 17:38:43 +0300 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <4A44D73F.2010408@freeswitch.org> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> Message-ID: <1246027123.4232.16.camel@dk-d820> > There doesn't seem to be any direct link between FreeSWITCH and Howler. > How does this benefit the project? Can we assume that Howler is "giving > something back"? Well, they're providing additional functionality which a lot of folk want, which requires substantial investment in money (visit sipro.com and have a look at the up-front license fees if you wish) and time (to get the thing working across the various platforms with the required licensing) and they've presumably taken a commercial risk so to do. Hats off to them: they've taken a risk, and the project benefits immensely from it. Expecting them to "give something back" in addition would be entirely unreasonable. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From d at d-man.org Fri Jun 26 08:01:38 2009 From: d at d-man.org (Darren Schreiber) Date: Fri, 26 Jun 2009 08:01:38 -0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com><22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com><8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com><39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> Message-ID: I can add a field to allow you to adjust the heartbeat on either channel if that's necessary. Right now you are right, it's using the global setting always. Is this important? -----Original Message----- From: Dome Charoenyost [mailto:dome at tel.co.th] Sent: Thursday, June 25, 2009 7:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway 2009/6/26 Michael Jerris : > I said to just add the set import=nibble_rate, your re-setting it for > no reason (and getting rid of the change that should have helped) by > your import=nibble_account line I test it agin. import work. nibble can see nibble_rate , nibble_account in channel but i can't change nibble heratbeat so nibble use default heartbeat. Dome C. > Mike > On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: > > Just test. > i use javascript > > ?????? session.execute("set", "import=nibble_rate"); > ?????? session.execute("set", "import=nibble_account"); > ?????? session.execute("bridge", > "{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=083 > 8833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); > > when call connected nibble do nothing? i found heartbeat > > mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! > when call disconnect nibble update amont. > mod_nibblebill.c:478 Billing 16 secs > > I think nibble still not found variable channel. > > Let's me share more information > > I want to use nibblebill for callingcard. (i have develop billing by > myself). i plan to use javascript connect to ODBC when customer call > my script query balance and say. > and then i loop for get destination (my customer want to dial many number). > when i got number my script query > gateway from DB.? i have 3 route and order by cost. > First plan i use > session.execute("bridge", > "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/xxxx > @provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/exte > rnal/xxxx at provder2"); i modify nibblebill for match provider with my > billing. > this case still fail. > > now i try > > ??? if > (session.ready()){ > ??????? s = new > Session("{absolute_codec_string='GSM,G729'}sofia/external/xxx at provider1" > > } > ??? if > (s.ready()){ > ??????? session.execute("set", > "nibble_rate=2.5"); > ??????? session.execute("set", > "nibble_account="+acaller); > ??????? session.execute("set", > "hangup_after_bridge=false"); > ??????? session.execute("set", > "provider_id="+dialprovider_id[1]); > > bridge(session,s); > ??? } > > and check hangup cause before try other provider. > > > > Please guide me it's right way or not ? > > > Dome C. > > > 2009/6/26 Darren Schreiber >> >> Did this work? Would love an update on this error/issue. >> ________________________________ >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Wednesday, June 24, 2009 8:15 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >> >> try adding >> before the >> bridge and report back results. >> Mike >> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >> >> Dear All, >> >> Look like nibblebill does't work with multiple gatreway. >> I try >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/extern >> al/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734 >> 000 at 202.xxx.xxx.xxx> >> >> nibblebill not found nibble_rate >> >> But >> ??????? >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203 >> .xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> Work fine >> >> What's difference from set application and []? ? >> >> Best Regards. >> Dome C. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Jun 26 08:08:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 26 Jun 2009 10:08:45 -0500 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <191c3a030906260802h790535cal4009d32225abee2f@mail.gmail.com> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> <1246027123.4232.16.camel@dk-d820> <191c3a030906260802h790535cal4009d32225abee2f@mail.gmail.com> Message-ID: <191c3a030906260808n684800afj7bbd8258354195c7@mail.gmail.com> Congrats to them for taking the risk. Just as an aside, just to let everyone know, I am taking the same risk. We have invested in the same deal from sipro and will shortly be offering an official FreeSWITCH g729 module in the very near future. This, of course, does benifit the project directly, because it gives a revenue source to help fund the project where naturally a 3rd party module mostly benifits users of FreeSWITCH which is, of course, not a bad thing. We might have had it sooner but we've been very busy making FreeSWITCH itself and it's a matter of first things first. So, again congrats guys, I hope your hardware codec boards sell and you leave a little room for us on the software side. On Jun 26, 2009 9:48 AM, "David Knell" wrote: > There doesn't seem to be any direct link between FreeSWITCH and Howler. > How does this benefit ... Well, they're providing additional functionality which a lot of folk want, which requires substantial investment in money (visit sipro.com and have a look at the up-front license fees if you wish) and time (to get the thing working across the various platforms with the required licensing) and they've presumably taken a commercial risk so to do. Hats off to them: they've taken a risk, and the project benefits immensely from it. Expecting them to "give something back" in addition would be entirely unreasonable. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lis... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/ddabec74/attachment.html From brian at freeswitch.org Fri Jun 26 08:13:32 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 10:13:32 -0500 Subject: [Freeswitch-users] Bug reports Message-ID: FreeSWITCHers, We have written an extensive guide on posting bugs to jira. Over the past few weeks everyone has been a little lax in posting the correct info the first time. The questions we ask are NOT optional to fill out.. You must fill out every question on the list and "doesn't apply" or "n/a" are not acceptable answers. So help us help you fix your issues that you report. Remember provide all the details even if you think its not relevant to the issue at hand because even the smallest detail can help me reproduce your issue or result in me pulling my hair out and going crazy. :P http://wiki.freeswitch.org/wiki/Reporting_Bugs Also I would like to solicit volunteers to help manage, track and collect info for any Jira's please contact me... I have asked a few times but very few expressed interest in helping. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/9d17c269/attachment.html From vhatz at kinetix.gr Fri Jun 26 08:20:22 2009 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Fri, 26 Jun 2009 18:20:22 +0300 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <1246027123.4232.16.camel@dk-d820> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> <1246027123.4232.16.camel@dk-d820> Message-ID: <4A44E736.9080501@kinetix.gr> I have to agree with David here. A G729 codec benefits the project immensely as it allows Freeswitch users to use the most widespread commercial codec. Best regards, Vlasis Hatzistavrou Kinetix Tele.com Hellas Ltd. Monastiriou 9 & Enotikon 54627 Thessaloniki Greece Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: vhatz at kinetix.gr http://www.kinetixtele.com David Knell wrote: >> There doesn't seem to be any direct link between FreeSWITCH and Howler. >> How does this benefit the project? Can we assume that Howler is "giving >> something back"? > > Well, they're providing additional functionality which a lot of folk > want, which requires substantial investment in money (visit sipro.com > and have a look at the up-front license fees if you wish) and time (to > get the thing working across the various platforms with the required > licensing) and they've presumably taken a commercial risk so to do. > > Hats off to them: they've taken a risk, and the project benefits > immensely from it. Expecting them to "give something back" in addition > would be entirely unreasonable. > > --Dave > From chris.chen2004 at gmail.com Fri Jun 26 08:26:16 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 26 Jun 2009 11:26:16 -0400 Subject: [Freeswitch-users] Bug reports In-Reply-To: References: Message-ID: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> Brian, I would like to be one of the volunteers helping to report issues. Chris On Fri, Jun 26, 2009 at 11:13 AM, Brian West wrote: > FreeSWITCHers, > > We have written an extensive guide on posting bugs to jira. Over the past > few weeks everyone has been a little lax in posting the correct info the > first time. The questions we ask are NOT optional to fill out.. You must > fill out every question on the list and "doesn't apply" or "n/a" are not > acceptable answers. So help us help you fix your issues that you report. > Remember provide all the details even if you think its not relevant to the > issue at hand because even the smallest detail can help me reproduce your > issue or result in me pulling my hair out and going crazy. :P > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also I would like to solicit volunteers to help manage, track and collect > info for any Jira's please contact me... I have asked a few times but very > few expressed interest in helping. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/6847f685/attachment.html From dome at tel.co.th Fri Jun 26 08:28:07 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 22:28:07 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> <24215893.post@talk.nabble.com> Message-ID: <8ccbff060906260828s368ba86dp38bd8f25c99e6f3e@mail.gmail.com> 2009/6/26 Harmeet Singh : > Just add the prefix like - > > > > BTW: Never give your real prefix. Anybody can use it to send traffic to your > trunk with that prefix and eat away your balance. ACL can help :) > > On Fri, Jun 26, 2009 at 3:05 AM, Edmar Cruz wrote: >> >> Yup your suggestions works... But I want my to have a prefix for the >> second >> bridge >> >> >> >> >> >> Dome Charoenyost wrote: >> > >> > Or Try pipe >> > ? >> > if you want to ring all. Try comma >> > ? >> > >> > >> > 2009/6/26 Edmar Cruz : >> >> >> >> >> >> ? >> >> ? ? >> >> ? ? ? >> >> ? ? ?> >> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >> ? ? ?> >> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >> ? ? >> >> ? ? ?> >> data="sofia/default/$1 at 116.80.80.101"/> >> >> ? ? >> >> >> >> ? ? >> >> ? ? ? >> >> ? ? ?> >> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >> ? ? ?> >> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >> >> >> ? ? ?> >> data="sofia/default/$1 at 116.80.80.102"/> >> >> ? ? >> >> >> >> ? >> >> >> >> >> >> I try what you said still not working... >> >> >> >> >> >> Dome Charoenyost wrote: >> >>> >> >>> May be need >> >>> >> >>> before first bridge >> >>> >> >>> >> >>> Dome C. >> >>> 2009/6/26 Edmar Cruz : >> >>>> >> >>>> >> >>>> ? >> >>>> ? ? >> >>>> ? ? ? >> >>>> ? ? ?> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >>>> ? ? ?> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >>>> ? ? ?--> >> >>>> ? ? ?> >>>> data="sofia/default/$1 at 116.80.80.101"/> >> >>>> ? ? >> >>>> >> >>>> ? ? >> >>>> ? ? ? >> >>>> ? ? ?> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >>>> ? ? ?> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >>>> ? ? ?--> >> >>>> ? ? ?> >>>> data="sofia/default/$1 at 116.80.80.102"/> >> >>>> ? ? >> >>>> >> >>>> ? >> >>>> >> >>>> >> >>>> >> >>>> Is this correct for multiple gateways? When I try this the first >> >>>> gateway >> >>>> works but the second gateway does not work? >> >>>> >> >>>> >> >>>> What is the solution for this can u help me? >> >>>> >> >>>> >> >>>> Thanks >> >>>> >> >>>> -- >> >>>> View this message in context: >> >>>> >> >>>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> Freeswitch-users mailing list >> >>>> Freeswitch-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> _______________________________________________ >> >>> Freeswitch-users mailing list >> >>> Freeswitch-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >>> >> >> >> >> -- >> >> View this message in context: >> >> >> >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215893.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dome at tel.co.th Fri Jun 26 08:26:59 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 22:26:59 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> <8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com> <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> Message-ID: <8ccbff060906260826g46c565c7sbe820d0b60cb7dd1@mail.gmail.com> 2009/6/26 Darren Schreiber : > I can add a field to allow you to adjust the heartbeat on either channel if > that's necessary. Right now you are right, it's using the global setting > always. > > Is this important? Yes important for me. and early media also :) 90% mobile in thailand use Music Ringback Tone service. if i disable ignore early media caller happy (heard music) but nibblebill start counting ;( Dome C. > > -----Original Message----- > From: Dome Charoenyost [mailto:dome at tel.co.th] > Sent: Thursday, June 25, 2009 7:39 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway > > 2009/6/26 Michael Jerris : >> I said to just add the set import=nibble_rate, your re-setting it for >> no reason (and getting rid of the change that should have helped) by >> your import=nibble_account line > I test it agin. > import work. ?nibble can see nibble_rate , nibble_account in channel but ?i > can't ?change nibble heratbeat ?so nibble use default heartbeat. > > > Dome C. > >> Mike >> On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: >> >> Just test. >> i use javascript >> >> ?????? session.execute("set", "import=nibble_rate"); >> ?????? session.execute("set", "import=nibble_account"); >> ?????? session.execute("bridge", >> "{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=083 >> 8833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); >> >> when call connected nibble do nothing? i found heartbeat >> >> mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! >> when call disconnect nibble update amont. >> mod_nibblebill.c:478 Billing 16 secs >> >> I think nibble still not found variable channel. >> >> Let's me share more information >> >> I want to use nibblebill for callingcard. (i have develop billing by >> myself). i plan to use javascript connect to ODBC when customer call >> my script query balance and say. >> and then i loop for get destination (my customer want to dial many > number). >> when i got number my script query >> gateway from DB.? i have 3 route and order by cost. >> First plan i use >> session.execute("bridge", >> "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/xxxx >> @provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/exte >> rnal/xxxx at provder2"); i modify nibblebill for match provider with my >> billing. >> this case still fail. >> >> now i try >> >> ??? if >> (session.ready()){ >> ??????? s = new >> Session("{absolute_codec_string='GSM,G729'}sofia/external/xxx at provider1" >> >> } >> ??? if >> (s.ready()){ >> ??????? session.execute("set", >> "nibble_rate=2.5"); >> ??????? session.execute("set", >> "nibble_account="+acaller); >> ??????? session.execute("set", >> "hangup_after_bridge=false"); >> ??????? session.execute("set", >> "provider_id="+dialprovider_id[1]); >> >> bridge(session,s); >> ??? } >> >> and check hangup cause before try other provider. >> >> >> >> Please guide me it's right way or not ? >> >> >> Dome C. >> >> >> 2009/6/26 Darren Schreiber >>> >>> Did this work? Would love an update on this error/issue. >>> ________________________________ >>> From: Michael Jerris [mailto:mike at jerris.com] >>> Sent: Wednesday, June 24, 2009 8:15 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >>> >>> try adding >>> before the >>> bridge and report back results. >>> Mike >>> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >>> >>> Dear All, >>> >>> Look like nibblebill does't work with multiple gatreway. >>> I try >>> ??????? >> data="nibble_account=0838833133"/> >>> ??????? >> data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/extern >>> al/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734 >>> 000 at 202.xxx.xxx.xxx> >>> >>> nibblebill not found nibble_rate >>> >>> But >>> ??????? >>> ??????? >> data="nibble_account=0838833133"/> >>> ??????? >> data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203 >>> .xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >>> >>> Work fine >>> >>> What's difference from set application and []? ? >>> >>> Best Regards. >>> Dome C. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jay.fenton at howlertech.com Fri Jun 26 08:28:13 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Fri, 26 Jun 2009 17:28:13 +0200 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <1246027123.4232.16.camel@dk-d820> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> <1246027123.4232.16.camel@dk-d820> Message-ID: <2825AB6A-55CB-4C19-A5E3-67D3A865D686@howlertech.com> Hey David, >> There doesn't seem to be any direct link between FreeSWITCH and >> Howler. >> How does this benefit the project? Can we assume that Howler is >> "giving >> something back"? > > Well, they're providing additional functionality which a lot of folk > want, which requires substantial investment in money (visit sipro.com > and have a look at the up-front license fees if you wish) and time (to > get the thing working across the various platforms with the required > licensing) and they've presumably taken a commercial risk so to do. > > Hats off to them: they've taken a risk, and the project benefits > immensely from it. Expecting them to "give something back" in > addition > would be entirely unreasonable. You're spot on on the somewhat extortionate up-front (and continuing royalty) fees, and there is substantial risk in us doing this which we hope will be mitigated by the rising interest in FreeSWITCH as a whole (a project which we've been huge evangelists of since the beginning). We hope that adding patent encumbered codecs like G.729 to FreeSWITCH will ultimately benefit the project and cause more people to be able to transition towards it from other platforms, building on the fantastic platform that Anthony and the rest of the gang have somehow managed to develop for free! Our initial feedback has been very promising, with a number of customers at least in the UK (including one major VoIP provider) that we've spoken to directly now seriously considering a migration to FreeSWITCH. We hope that the small per channel charge is not too onerous - we've tried to make it as cheap as possible by introducing the floating licenses (and including all support and maintenance upgrades free of charge), but there are very real costs here in royalties and development - how you guys do it mostly for free is beyond me! We have, of course, spoken to the FreeSWITCH team and offered to collaborate on this project (as they've also been working on this for a while) and we hope that we can join forces on it soon so that everyone benefits :) -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From mike at jerris.com Fri Jun 26 08:42:50 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 11:42:50 -0400 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <8ccbff060906260826g46c565c7sbe820d0b60cb7dd1@mail.gmail.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> <8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com> <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> <8ccbff060906260826g46c565c7sbe820d0b60cb7dd1@mail.gmail.com> Message-ID: <63B6DBF6-C57B-43CB-893B-73EA6EC8BC57@jerris.com> if you want to import multiple vars you can set import to a comma separated list of vars. This should solve the problem in the short term, the question for Darren is, do you think we should make the mod read from the a or b channel for those vars. Mike On Jun 26, 2009, at 11:26 AM, Dome Charoenyost wrote: > 2009/6/26 Darren Schreiber : >> I can add a field to allow you to adjust the heartbeat on either >> channel if >> that's necessary. Right now you are right, it's using the global >> setting >> always. >> >> Is this important? > Yes important for me. and early media also :) > > 90% mobile in thailand use Music Ringback Tone service. if i disable > ignore early media caller happy (heard music) > but nibblebill start counting ;( > > > > > Dome C. > > >> >> -----Original Message----- >> From: Dome Charoenyost [mailto:dome at tel.co.th] >> Sent: Thursday, June 25, 2009 7:39 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >> >> 2009/6/26 Michael Jerris : >>> I said to just add the set import=nibble_rate, your re-setting it >>> for >>> no reason (and getting rid of the change that should have helped) by >>> your import=nibble_account line >> I test it agin. >> import work. nibble can see nibble_rate , nibble_account in >> channel but i >> can't change nibble heratbeat so nibble use default heartbeat. >> >> >> Dome C. >> >>> Mike >>> On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: >>> >>> Just test. >>> i use javascript >>> >>> session.execute("set", "import=nibble_rate"); >>> session.execute("set", "import=nibble_account"); >>> session.execute("bridge", >>> "{absolute_codec_string='GSM,G729'} >>> [nibble_rate=0.5,nibble_account=083 >>> 8833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); >>> >>> when call connected nibble do nothing i found heartbeat >>> >>> mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! >>> when call disconnect nibble update amont. >>> mod_nibblebill.c:478 Billing 16 secs >>> >>> I think nibble still not found variable channel. >>> >>> Let's me share more information >>> >>> I want to use nibblebill for callingcard. (i have develop billing by >>> myself). i plan to use javascript connect to ODBC when customer call >>> my script query balance and say. >>> and then i loop for get destination (my customer want to dial many >> number). >>> when i got number my script query >>> gateway from DB. i have 3 route and order by cost. >>> First plan i use >>> session.execute("bridge", >>> "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/ >>> xxxx >>> @provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/ >>> exte >>> rnal/xxxx at provder2"); i modify nibblebill for match provider with my >>> billing. >>> this case still fail. >>> >>> now i try >>> >>> if >>> (session.ready()){ >>> s = new >>> Session("{absolute_codec_string='GSM,G729'}sofia/external/ >>> xxx at provider1" >>> >>> } >>> if >>> (s.ready()){ >>> session.execute("set", >>> "nibble_rate=2.5"); >>> session.execute("set", >>> "nibble_account="+acaller); >>> session.execute("set", >>> "hangup_after_bridge=false"); >>> session.execute("set", >>> "provider_id="+dialprovider_id[1]); >>> >>> bridge(session,s); >>> } >>> >>> and check hangup cause before try other provider. >>> >>> >>> >>> Please guide me it's right way or not ? >>> >>> >>> Dome C. >>> >>> >>> 2009/6/26 Darren Schreiber >>>> >>>> Did this work? Would love an update on this error/issue. >>>> ________________________________ >>>> From: Michael Jerris [mailto:mike at jerris.com] >>>> Sent: Wednesday, June 24, 2009 8:15 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >>>> >>>> try adding >>>> before the >>>> bridge and report back results. >>>> Mike >>>> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >>>> >>>> Dear All, >>>> >>>> Look like nibblebill does't work with multiple gatreway. >>>> I try >>>> >>> data="nibble_account=0838833133"/> >>>> >>> data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/ >>>> extern >>>> al/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/ >>>> 6626734 >>>> 000 at 202.xxx.xxx.xxx> >>>> >>>> nibblebill not found nibble_rate >>>> >>>> But >>>> >>>> >>> data="nibble_account=0838833133"/> >>>> >>> data="{absolute_codec_string='GSM,G729'}sofia/external/ >>>> 6626734000 at 203 >>>> .xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >>>> >>>> Work fine >>>> >>>> What's difference from set application and [] ? >>>> >>>> Best Regards. >>>> Dome C. >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>> ers >>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>> ers >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> rs >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> rs >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mail at willboyce.com Fri Jun 26 08:43:27 2009 From: mail at willboyce.com (Will Boyce) Date: Fri, 26 Jun 2009 10:43:27 -0500 (CDT) Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <10974959.701246030807467.JavaMail.SYSTEM@man-00108> Message-ID: <8044499.721246030958844.JavaMail.SYSTEM@man-00108> Hey David, >> There doesn't seem to be any direct link between FreeSWITCH and >> Howler. >> How does this benefit the project? Can we assume that Howler is >> "giving >> something back"? > > Well, they're providing additional functionality which a lot of folk > want, which requires substantial investment in money (visit sipro.com > and have a look at the up-front license fees if you wish) and time (to > get the thing working across the various platforms with the required > licensing) and they've presumably taken a commercial risk so to do. > > Hats off to them: they've taken a risk, and the project benefits > immensely from it. Expecting them to "give something back" in > addition > would be entirely unreasonable. You're spot on on the somewhat extortionate up-front (and continuing royalty) fees, and there is substantial risk in us doing this which we hope will be mitigated by the rising interest in FreeSWITCH as a whole (a project which we've been huge evangelists of since the beginning). We hope that adding patent encumbered codecs like G.729 to FreeSWITCH will ultimately benefit the project and cause more people to be able to transition towards it from other platforms, building on the fantastic platform that Anthony and the rest of the gang have somehow managed to develop for free! Our initial feedback has been very promising, with a number of customers at least in the UK (including one major VoIP provider) that we've spoken to directly now seriously considering a migration to FreeSWITCH. We hope that the small per channel charge is not too onerous - we've tried to make it as cheap as possible by introducing the floating licenses (and including all support and maintenance upgrades free of charge), but there are very real costs here in royalties and development - how you guys do it mostly for free is beyond me! We have, of course, spoken to the FreeSWITCH team and offered to collaborate on this project (as they've also been working on this for a while) and we hope that we can join forces on it soon so that everyone benefits :) -- Regards, Jay Fenton, CTO c/o Will Boyce Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton From mike at jerris.com Fri Jun 26 09:03:10 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 12:03:10 -0400 Subject: [Freeswitch-users] From Asterisk to Freeswitch In-Reply-To: <8ccbff060906252345w1aa7c4a1u66f1c2af752dea20@mail.gmail.com> References: <8ccbff060906252345w1aa7c4a1u66f1c2af752dea20@mail.gmail.com> Message-ID: <8C494316-C254-42F5-BF69-ED687AF2E6E0@jerris.com> most of the information about fifo is : http://wiki.freeswitch.org/wiki/Mod_fifo Mike On Jun 26, 2009, at 2:45 AM, Dome Charoenyost wrote: > Dear All, > > I'm asterisk developer(I have some code in Asterisk) . After 3 > weeks with freeswich nothing to say. now i'm move all callingcard , > wholesale platfrom to Freeswitch. I'm very happy with bridge , > nibblebill. after finish this job i'll test FS PBX feature. i think > it's easy to do Hosted IPPBX. But i want to know more about mod_fifo > Can someone tell me about mod_fifo compare with asterisk app_queue. > i'm talking about annouce , priority agent > From mike at jerris.com Fri Jun 26 09:05:52 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 12:05:52 -0400 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> Message-ID: <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> I would pre-adjust the volume of the soundfile with sox instead of doing it real time. Mike On Jun 26, 2009, at 7:38 AM, Dennis wrote: > hi, > > we want to use uuid_displace with mux to playback a soundfile to a > bridged uuid, so that this uuid can hear the other side talk AND hear > the soundfile (whispering). > > is there an option we can set, for defining the loudness level of the > soundfile? in our tests the soundfile was way to loud, so that it was > nearly impossible to hear the other side talk, while the soundfile was > playing. > > we tried "uuid_displace uuid start /path/to/soundfile/soundfile.wav 0 > mux 0.3", so that the loudness of soundfile only would be 30% - but > this does not work. > From solko at gcdf.pl Fri Jun 26 09:14:10 2009 From: solko at gcdf.pl (Szymon Olko) Date: Fri, 26 Jun 2009 18:14:10 +0200 Subject: [Freeswitch-users] Bug reports In-Reply-To: References: Message-ID: <4A44F3D2.10901@gcdf.pl> Brian West pisze: > FreeSWITCHers, > > We have written an extensive guide on posting bugs to jira. Over the > past few weeks everyone has been a little lax in posting the correct > info the first time. The questions we ask are NOT optional to fill > out.. You must fill out every question on the list and "doesn't apply" > or "n/a" are not acceptable answers. So help us help you fix your > issues that you report. Remember provide all the details even if you > think its not relevant to the issue at hand because even the smallest > detail can help me reproduce your issue or result in me pulling my hair > out and going crazy. :P > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also I would like to solicit volunteers to help manage, track and > collect info for any Jira's please contact me... I have asked a few > times but very few expressed interest in helping. > I agree that those information are needed for developer, I will fill them all. What would make it easy for me if there was a "clone" option to make new ticked based on old one but with ability to change information. Most of things you ask are the same in all my tickes. I would just need to put new bug description and change only few items. If this is not possible then maybe jira has something like "ticket template" which I can fill once and then make all tickets base on it. Regards Szymon PS: Wiki page is great > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Fri Jun 26 09:37:22 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 26 Jun 2009 19:37:22 +0300 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <8044499.721246030958844.JavaMail.SYSTEM@man-00108> References: <8044499.721246030958844.JavaMail.SYSTEM@man-00108> Message-ID: <1246034242.4232.25.camel@dk-d820> On Fri, 2009-06-26 at 10:43 -0500, Will Boyce wrote: > Hey David, [duplicate post snipped] -- don't suppose your product includes an echo canceller, does it ;-) Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From shiyanov at gmail.com Fri Jun 26 10:25:50 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Fri, 26 Jun 2009 21:25:50 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge Message-ID: Hello! I got a problem with one way audio, symptoms are: firstly play audio file to channel A (A is hears sound) secondly bridge channel B with A (A doesn't hear B). Environment: - no NAT - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch - dialplan: - Call routing scheme: user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc Exact description what's going on is: user A -> FS -(bridge)-> my B2BUA Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) user to extension "Local_Extension_from_SP". This should create a new call to user B. As a result - A doesn't hear B, but B- is OK. On the contrary, if a call is routed (by B2BUA) to the "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - everything is OK. What I've tried: - set parameter "inbound-proxy-media" to "true" in Sofia profile - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile Nothing helps. Any help or thoughts would be MUCH appreciated! Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/7e19e993/attachment.html From msc at freeswitch.org Fri Jun 26 10:46:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 10:46:34 -0700 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> Message-ID: <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> On Fri, Jun 26, 2009 at 9:05 AM, Michael Jerris wrote: > I would pre-adjust the volume of the soundfile with sox instead of > doing it real time. > > Mike > I have to agree with Mike here. Sox is awesome for this kind of thing and disk space is way more plentiful than CPU power. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/a51a2cd1/attachment.html From msc at freeswitch.org Fri Jun 26 10:50:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 10:50:10 -0700 Subject: [Freeswitch-users] Service_not_implemented for mobile phones? In-Reply-To: <24217116.post@talk.nabble.com> References: <24217116.post@talk.nabble.com> Message-ID: <87f2f3b90906261050q6ff09ac2ka0a6e06f3c8a924d@mail.gmail.com> Edmar, I strongly recommend you review the troubleshooting page on the wiki. No one will be able to help you without more information about your issue. http://wiki.freeswitch.org/wiki/Reporting_Bugs Be sure to use pastebin and capture the debug output from the FS command line and also paste your relevant configuration files. -MC On Fri, Jun 26, 2009 at 2:03 AM, Edmar Cruz wrote: > > Hi, > > I receive an error message service not implemented sometimes when calling > a mobile phone number but sometimes it works. What maybe rhe cause of this > error? I already installed zfone, perfectly connect to two freeswitch and > the one issue I got today is these can you help me guys? > > > Thanks > -- > View this message in context: > http://www.nabble.com/Service_not_implemented-for-mobile-phones--tp24217116p24217116.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/d5aa1942/attachment-0001.html From brian at freeswitch.org Fri Jun 26 10:52:52 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 12:52:52 -0500 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> Message-ID: <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> Are you playing a stereo file by chance? /b On Jun 26, 2009, at 12:46 PM, Michael Collins wrote: > > > On Fri, Jun 26, 2009 at 9:05 AM, Michael Jerris > wrote: > I would pre-adjust the volume of the soundfile with sox instead of > doing it real time. > > Mike > > I have to agree with Mike here. Sox is awesome for this kind of > thing and disk space is way more plentiful than CPU power. > > -MC > _______ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/98b6efed/attachment.html From brian at freeswitch.org Fri Jun 26 10:54:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 12:54:00 -0500 Subject: [Freeswitch-users] Bug reports In-Reply-To: <4A44F3D2.10901@gcdf.pl> References: <4A44F3D2.10901@gcdf.pl> Message-ID: <58A2C686-BEE7-4F17-825C-FCFE922F07F5@freeswitch.org> Well I'm going to try my best to reproduce issues on demand but sometimes I just can't, other times the reporter stops responding and I have no choice but to close the issue out. :) /b On Jun 26, 2009, at 11:14 AM, Szymon Olko wrote: > I agree that those information are needed for developer, I will fill > them all. > > What would make it easy for me if there was a "clone" option to make > new ticked based on old one but with ability to change > information. Most of things you ask are the same in all my tickes. I > would just need to put new bug description and change only > few items. > If this is not possible then maybe jira has something like "ticket > template" which I can fill once and then make all tickets base > on it. > > Regards > Szymon > > PS: Wiki page is great From brian at freeswitch.org Fri Jun 26 10:55:42 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 12:55:42 -0500 Subject: [Freeswitch-users] Bug reports In-Reply-To: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> References: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> Message-ID: It involves more than just reporting... it involves trying to reproduce the issue or clarify the steps to reproduce an issue.. I have been thru some issues that sound crazy hard to reproduce... like stand on your head .. dial 1234, spin around dial 456, scream into the phone... dance a little jig... then hangup... I know it sound silly but there have been bugs that are so complex to reproduce you end up pulling your hair out :P Granted they are few and far between and mostly corner cases. /b On Jun 26, 2009, at 10:26 AM, Chris Chen wrote: > Brian, I would like to be one of the volunteers helping to report > issues. > > Chris From shiyanov at gmail.com Fri Jun 26 10:59:38 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Fri, 26 Jun 2009 21:59:38 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: Updates: 1. One-way audio is in 95% tries. But how the rest 5% works?? 2. Strange FS logging after the channels are bridged (user A talk to user B) 2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at 192.168.147.1 entering state [ready] 2009-06-26 02:16:07 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 192.168.147.130 s=FreeSWITCH c=IN IP4 192.168.147.130 t=0 0 m=audio 31134 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 0 RTP/AVP 34 a=rtpmap:34 H263/90000 2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1000000000 at 192.168.147.130:5060 entering state [ready] freeswitch at localhost.localdomain> 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/ 1005 at uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:17:09 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 freeswitch at localhost.localdomain> show calls API CALL [show(calls)] output: created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2009-06-26 02:16:05,1245968165,switch_ivr_multi_threaded_bridge,1005,1005,inbound1000000000,sofia/external/ 1005 at uat.pbx.starpoundtech.net ,4fa86434-b542-4066-99af-5924c78ddab7,1005,1005, 1000000000 at 192.168.147.130:5060,sofia/external/ 1000000000 at 192.168.147.130:5060,73df8735-fee2-464d-aec0-fda886ba2cba 2009-06-26 02:16:07,1245968167,switch_ivr_multi_threaded_bridge,1005,1005,1001,sofia/external/ 1005 at 192.168.147.1 ,1c2c5f6d-669f-4432-ad04-35a64dbc8a14,1005,1005,sip:1001 at 192.168.147.1:5060 ;fs_nat=yes,sofia/doublenat5090/sip:1001 at 192.168.147.1:5060 ;fs_nat=yes,66895f68-70bf-410a-bff7-cda9549c102d 2 total. freeswitch at localhost.localdomain> 2009-06-26 02:18:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/ 1005 at uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:18:10 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:18:10 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-26 02:19:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:19:08 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:19:08 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Artem On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov wrote: > Hello! > > I got a problem with one way audio, symptoms are: > firstly play audio file to channel A (A is hears sound) > secondly bridge channel B with A (A doesn't hear B). > > Environment: > - no NAT > - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of > them- no audio, Wireshark shows that there is no RTP-flow to A from > FreeSwitch > - dialplan: > > > > > > > > > > > > > > > > expression="^${caller_id_number}$"> > > data="transfer_ringback=${us-ring}"/> > > > data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > data="sip_h_X-SPFrom="e;${sip_from_user}"e;<${sip_from_uri}>"/> > data="sip_h_X-SPTo=<${sip_to_uri}>"/> > data="sip_h_X-SPCallId=${sip_call_id}"/> > data="sofia/external/${orgname}send2voicemail@ > $${starpound_sip_app_server}"/> > > > - Call routing scheme: > user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc > Exact description what's going on is: > user A -> FS -(bridge)-> my B2BUA > Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to > extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) > user to extension "Local_Extension_from_SP". This should create a new call > to user B. As a result - A doesn't hear B, but B- is OK. > On the contrary, if a call is routed (by B2BUA) to the > "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - > everything is OK. > > > What I've tried: > - set parameter "inbound-proxy-media" to "true" in Sofia profile > - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile > Nothing helps. > > > Any help or thoughts would be MUCH appreciated! > Artem > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/f49f68f1/attachment.html From brian at freeswitch.org Fri Jun 26 11:03:40 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 13:03:40 -0500 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk also... due to the lines below. /b On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 > 192.168.147.130 > s=FreeSWITCH From Mailings at kh-dev.de Fri Jun 26 12:48:36 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 26 Jun 2009 21:48:36 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Message-ID: Hi all, I'm just writing a perl script as dialplan to learn how to handle freeswitch. Now I have the issue that my hangup hook won't be triggered after I intercepted a call. A "normal" hangup triggers the function. Does anybody have a hint how to get the hangup hook triggered? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/503b4419/attachment.html From brian at freeswitch.org Fri Jun 26 13:06:52 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 15:06:52 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: Message-ID: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> Depends what side of the call was the hangup hook on? /b On Jun 26, 2009, at 2:48 PM, Klaus Hochlehnert wrote: > Hi all, > > I?m just writing a perl script as dialplan to learn how to handle > freeswitch. > > Now I have the issue that my hangup hook won?t be triggered after I > intercepted a call. > A ?normal? hangup triggers the function. > > Does anybody have a hint how to get the hangup hook triggered? > > Thanks, Klaus > ______________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/ece4df34/attachment.html From Mailings at kh-dev.de Fri Jun 26 13:22:14 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 26 Jun 2009 22:22:14 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> Message-ID: Actually one of my first actions in the script is $session->setHangupHook('on_hangup'); When a call comes in the hook is set and working. The second time the script is called when I try to intercept. As it's the same script there's also the function setHangupHook called. That's what I've currently done. How can I set up the hook for the "new" bridge? Or is there a possibility to set a global hook? Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 26, 2009 10:07 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Depends what side of the call was the hangup hook on? /b On Jun 26, 2009, at 2:48 PM, Klaus Hochlehnert wrote: Hi all, I'm just writing a perl script as dialplan to learn how to handle freeswitch. Now I have the issue that my hangup hook won't be triggered after I intercepted a call. A "normal" hangup triggers the function. Does anybody have a hint how to get the hangup hook triggered? Thanks, Klaus ______________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/c326f510/attachment.html From brian at freeswitch.org Fri Jun 26 13:27:32 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 15:27:32 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> Message-ID: <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> well in your case I suspect your intercepting the leg of the call without the hook on it. /b On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: > Actually one of my first actions in the script is > $session->setHangupHook('on_hangup'); > > When a call comes in the hook is set and working. > > The second time the script is called when I try to intercept. As > it?s the same script there?s also the function setHangupHook called. > That?s what I?ve currently done. > > How can I set up the hook for the ?new? bridge? > Or is there a possibility to set a global hook? > > Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/ba8c9d6c/attachment-0001.html From Mailings at kh-dev.de Fri Jun 26 13:39:16 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 26 Jun 2009 22:39:16 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> Message-ID: Ok, and how can I ask the hook to come with me? If I understand this right... When a call comes in the hook is set on the a-leg and it rings on the b-leg. When I do an intercept I kill the ringing b-leg and the interceptor is now the "new" b-leg, right? I would assume that the "old" a-leg still has the hook on it or this wrong. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 26, 2009 10:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... well in your case I suspect your intercepting the leg of the call without the hook on it. /b On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: Actually one of my first actions in the script is $session->setHangupHook('on_hangup'); When a call comes in the hook is set and working. The second time the script is called when I try to intercept. As it's the same script there's also the function setHangupHook called. That's what I've currently done. How can I set up the hook for the "new" bridge? Or is there a possibility to set a global hook? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/1cee6d64/attachment.html From msc at freeswitch.org Fri Jun 26 13:56:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 13:56:43 -0700 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> Message-ID: <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> Can you paste in your script so we can see what is going on? -MC On Fri, Jun 26, 2009 at 1:39 PM, Klaus Hochlehnert wrote: > Ok, and how can I ask the hook to come with me? > > > > If I understand this right... > > When a call comes in the hook is set on the a-leg and it rings on the > b-leg. > > When I do an intercept I kill the ringing b-leg and the interceptor is now > the ?new? b-leg, right? > > I would assume that the ?old? a-leg still has the hook on it or this wrong. > > > > Thanks, Klaus > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Friday, June 26, 2009 10:28 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] hangup hook after intercept doesn't get > triggered... > > > > well in your case I suspect your intercepting the leg of the call without > the hook on it. > > > > /b > > > > On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: > > > > Actually one of my first actions in the script is > > $session->setHangupHook('on_hangup'); > > > > When a call comes in the hook is set and working. > > > > The second time the script is called when I try to intercept. As it?s the > same script there?s also the function setHangupHook called. > > That?s what I?ve currently done. > > > > How can I set up the hook for the ?new? bridge? > > Or is there a possibility to set a global hook? > > > > Thanks, Klaus > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/5da26bc9/attachment.html From john at feith.com Fri Jun 26 14:33:05 2009 From: john at feith.com (John Wehle) Date: Fri, 26 Jun 2009 17:33:05 -0400 (EDT) Subject: [Freeswitch-users] Accessing a global variable from lua Message-ID: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> How do you get a system variable from within a lua startup script? Specifically I want domain_name from vars.xml ... normally I'd use session:getVariable, however there is no session in this case. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From brian at freeswitch.org Fri Jun 26 14:37:48 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 16:37:48 -0500 Subject: [Freeswitch-users] Accessing a global variable from lua In-Reply-To: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> References: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> Message-ID: <60AC4243-2C80-4A0D-8B7E-F908D49A5BA7@freeswitch.org> You can execute global_getvar api call. /b On Jun 26, 2009, at 4:33 PM, John Wehle wrote: > How do you get a system variable from within a lua startup script? > Specifically I want domain_name from vars.xml ... normally I'd use > session:getVariable, however there is no session in this case. From msc at freeswitch.org Fri Jun 26 14:41:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 14:41:50 -0700 Subject: [Freeswitch-users] Accessing a global variable from lua In-Reply-To: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> References: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> Message-ID: <87f2f3b90906261441oe79997fod20c0ddb291d3984@mail.gmail.com> Use the API to execute global_getvar... http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls -MC On Fri, Jun 26, 2009 at 2:33 PM, John Wehle wrote: > How do you get a system variable from within a lua startup script? > Specifically I want domain_name from vars.xml ... normally I'd use > session:getVariable, however there is no session in this case. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/41c75cfa/attachment-0001.html From john at feith.com Fri Jun 26 15:13:12 2009 From: john at feith.com (John Wehle) Date: Fri, 26 Jun 2009 18:13:12 -0400 (EDT) Subject: [Freeswitch-users] Accessing a global variable from lua Message-ID: <200906262213.n5QMDCgh005662@jwlab.FEITH.COM> > You can execute global_getvar api call. Thanks ... I've updated the wiki. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From brian at freeswitch.org Fri Jun 26 15:31:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 17:31:19 -0500 Subject: [Freeswitch-users] Accessing a global variable from lua In-Reply-To: <200906262213.n5QMDCgh005662@jwlab.FEITH.COM> References: <200906262213.n5QMDCgh005662@jwlab.FEITH.COM> Message-ID: John, Also can you go over the few jira's you have left and see if we can resolve/knock out some of them.. I'm wanting to roll pre9 this weekend. Thanks, Brian On Jun 26, 2009, at 5:13 PM, John Wehle wrote: >> You can execute global_getvar api call. > > Thanks ... I've updated the wiki. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: > john at feith.com | > | John Wehle | Fax: 1-215-540-5495 > | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Mailings at kh-dev.de Fri Jun 26 16:16:31 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sat, 27 Jun 2009 01:16:31 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> Message-ID: Brian told me to open a jira, what I did now. But here's the script. It basically writes incoming calls to a database and removes them after hangup. When intercepting it also rewrites the destination of the call. Thanks, Klaus #!/usr/bin/perl use strict; use DBI; use POSIX qw(strftime); our $session; use constant { false => 0, true => 1, }; my $EmptyString = " "; # Used to delete regexp catches my $dbargs = {AutoCommit => 0, PrintError => 1}; my $dbh = DBI->connect("dbi:SQLite:dbname=/opt/freeswitch/db/dialplan_call_info.db", "", "", $dbargs); $session->setHangupHook('on_hangup'); logInfo("Hook set"); sub logString { my ($Level, $Msg) = @_; freeswitch::consoleLog("$Level", "$Msg\n"); } sub logDebug { my ($Msg) = @_; logString("DEBUG", ">>>>> $Msg"); } sub logInfo { my ($Msg) = @_; logString("INFO", ">>>>> $Msg"); } sub logNotice { my ($Msg) = @_; logString("NOTICE", ">>>>> $Msg"); } sub logWarning { my ($Msg) = @_; logString("WARNING", ">>>>> $Msg"); } sub logError { my ($Msg) = @_; logString("ERR", ">>>>> $Msg"); } sub logCritical { my ($Msg) = @_; logString("CRIT", ">>>>> $Msg"); } sub logAlert { my ($Msg) = @_; logString("ALERT", ">>>>> $Msg"); } # The idea of these functions is to allow for easy pull in of variables and then # automatically export any ones that have been changed when UPDATEV. # It will ensure you don't write to any non-imported variables, but as we are # using a hash we cannot prevent invalid reads. If you are really concerned about # this then you could use a specific read function which first checks to make sure # its defined in CLEAN_VARS before returning. my %VARS; my %CLEAN_VARS; # Takes one or more variables names to import in sub GETV { my @Arr = @_; foreach my $Var (@Arr) { $VARS{$Var} = $session->getVariable("$Var"); $CLEAN_VARS{$Var} = $VARS{$Var}; if (! defined $CLEAN_VARS{$Var}) { $CLEAN_VARS{$Var} = ""; } } } # Generally not called directly, but will set the variable to the value requested right away sub SETV { my ($Var, $Value) = @_; $session->setVariable("$Var", "$Value"); $VARS{$Var} = "$Value"; $CLEAN_VARS{$Var} = "$Value"; } # If we don't care about a variables value, but wan't to override it this will add it to the hash # so that when we write to it, we don't consider it a typo sub ADDV { my @Arr = @_; foreach my $Var(@Arr) { $CLEAN_VARS{$Var} = "123zzzzzZnzZZzz"; # Something definitely won't match $VARS{$Var} = ""; } } # Updates any changed variables sub UPDATEV { foreach my $Var (keys %VARS) { # Make sure there were no typos if (! defined $CLEAN_VARS{$Var}) { die "Warning a variable of: '$Var' was not found in CLEAN_VARS, did you forget to GET/ADD it first?"; } if ($VARS{$Var} ne $CLEAN_VARS{$Var}) { SETV($Var, $VARS{$Var}); } } } # Dump all variables sub DUMPV { foreach my $Var (sort keys %VARS) { logInfo("$Var = '" . $VARS{$Var} . "'"); } } sub CAN_ACCESS { my ($Req) = @_; if ($VARS{app_rights} eq "ALL" || $VARS{app_rights} =~ /#$Req#/) { return true; } else { return false; } } # Fetch some generic variables GETV("uuid", "base_dir", "domain", "app_rights", "de-ring", "outgoing_soundtouch_profile", "hold_music", "continue_on_fail"); GETV("destination_number", "caller_id_name", "caller_id_number", "effective_caller_id_name", "effective_caller_id_number"); GETV("network_addr", "hangup_after_bridge", "called_party_callgroup", "ringback", "transfer_ringback", "sip_exclude_contact"); GETV("call_timeout", "source", "sip_to_params", "presence_id", "dialed_user", "dialed_domain"); GETV("voicemail_authorized", "sip_authorized", "username", "accountcode", "sip_from_user", "sip_to_user"); # Set some defaults $VARS{hangup_after_bridge} = "true"; $VARS{ringback} = $VARS{'de-ring'}; $VARS{transfer_ringback} = $VARS{hold_music}; $VARS{sip_exclude_contact} = $VARS{network_addr}; $VARS{call_timeout} = "60"; $VARS{continue_on_fail} = "true"; # NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION UPDATEV(); sub bridgeCallInternally { my ($DestNr) = @_; if ("${DestNr}" == "21") { $VARS{call_timeout} = "15"; } UPDATEV(); $dbh->do("insert into current_calls (extension, uuid) values ('$DestNr', '$VARS{uuid}')"); $dbh->commit(); $session->execute("record_session","\${base_dir}/recordings/\${strftime(%Y-%m-%d-%H-%M-%S)}_\${destination_number}_\${caller_id_number}.wav"); $session->execute("bridge","user/${DestNr}\@$VARS{domain}"); } sub on_hangup { my $hup_session = shift; my $hup_cause = shift; logInfo("Hangup uuid: '" . $hup_session->{uuid} . "'"); logInfo("Hangup cause: '$hup_cause'"); $dbh->do("delete from current_calls where uuid = '" . $hup_session->{uuid} . "'"); $dbh->commit(); } # Internal numbers if ($VARS{destination_number} =~ /^(2[0-2])$/) { UPDATEV(); bridgeCallInternally($VARS{destination_number}); } # Intercept call if ($VARS{destination_number} =~ /^\*8(\d+)$/) { my $intercept_extension = ""; my $intercept_uuid = ""; my $sth = $dbh->prepare("select * from current_calls where extension = ?"); $sth->execute($1); while (my @data = $sth->fetchrow_array()) { $intercept_extension = $data[0]; $intercept_uuid = $data[1]; } logInfo("Intercept call from '$intercept_extension' - '$intercept_uuid'"); GETV("caller_id_number"); $dbh->do("update current_calls set extension = '$VARS{caller_id_number}' where uuid = '$intercept_uuid'"); $dbh->commit(); $session->answer(); $session->execute("intercept", "$intercept_uuid"); $session->execute("sleep", "1000"); } $dbh->disconnect(); return 1; From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, June 26, 2009 10:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Can you paste in your script so we can see what is going on? -MC On Fri, Jun 26, 2009 at 1:39 PM, Klaus Hochlehnert > wrote: Ok, and how can I ask the hook to come with me? If I understand this right... When a call comes in the hook is set on the a-leg and it rings on the b-leg. When I do an intercept I kill the ringing b-leg and the interceptor is now the "new" b-leg, right? I would assume that the "old" a-leg still has the hook on it or this wrong. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 26, 2009 10:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... well in your case I suspect your intercepting the leg of the call without the hook on it. /b On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: Actually one of my first actions in the script is $session->setHangupHook('on_hangup'); When a call comes in the hook is set and working. The second time the script is called when I try to intercept. As it's the same script there's also the function setHangupHook called. That's what I've currently done. How can I set up the hook for the "new" bridge? Or is there a possibility to set a global hook? Thanks, Klaus _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/32ccd95d/attachment-0001.html From brian at freeswitch.org Fri Jun 26 16:27:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 18:27:47 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> Message-ID: <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> Please open the jira and attach all this and the xml dialplan to execute it .. also the schema for the db would be helpful also. Thanks, /b On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote: > Brian told me to open a jira, what I did now. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/962bcd6b/attachment.html From Mailings at kh-dev.de Fri Jun 26 16:36:34 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sat, 27 Jun 2009 01:36:34 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> Message-ID: I used the Project "FS Scripts". Hope that's ok. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, June 27, 2009 1:28 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Please open the jira and attach all this and the xml dialplan to execute it .. also the schema for the db would be helpful also. Thanks, /b On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote: Brian told me to open a jira, what I did now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/aabf0f56/attachment.html From brian at freeswitch.org Fri Jun 26 16:40:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 18:40:38 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> Message-ID: <16FDA78B-5681-410D-AECA-5C50C9FD4A73@freeswitch.org> Its more FSCORE but i'll move it into the right place ;) /b On Jun 26, 2009, at 6:36 PM, Klaus Hochlehnert wrote: > I used the Project ?FS Scripts?. Hope that?s ok. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Saturday, June 27, 2009 1:28 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't > get triggered... > > Please open the jira and attach all this and the xml dialplan to > execute it .. also the schema for the db would be helpful also. > > Thanks, > /b > > On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote: > > > Brian told me to open a jira, what I did now. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/f2529776/attachment.html From jason at jasonjgw.net Fri Jun 26 17:02:14 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 10:02:14 +1000 Subject: [Freeswitch-users] Bug reports In-Reply-To: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> References: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> Message-ID: <20090627000214.GA7509@jdc.jasonjgw.net> Chris Chen wrote: > Brian, I would like to be one of the volunteers helping to report issues. That's great. We need more volunteers. For some FreeSWITCH users (of whom I am one), the user interface of Jira is an obstacle to reporting bugs via that mechanism, for accessibility reasons. I've had better luck with Bugzilla and other bug tracking systems, but Jira really seems to depend on Javascript for its fundamental operations, and that's a real barrier for me at the moment. From jason at jasonjgw.net Fri Jun 26 17:05:41 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 10:05:41 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <4A44AB08.1050503@gmail.com> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> Message-ID: <20090627000541.GB7509@jdc.jasonjgw.net> Tamas wrote: > Did you make bootstrap.sh and configure before compilation? Yes. This was a clean export from svn, built by running the Debian debuild tool, as in svn export to a temporary directory, followed by debuild (after changing the version number to make the package version unique). From brian at freeswitch.org Fri Jun 26 17:10:35 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 19:10:35 -0500 Subject: [Freeswitch-users] Bug reports In-Reply-To: <20090627000214.GA7509@jdc.jasonjgw.net> References: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> <20090627000214.GA7509@jdc.jasonjgw.net> Message-ID: Jason, In your case we'll gladly accept your bug reports via the mailing list for that exact reason. ;) /b On Fri, Jun 26, 2009 at 7:02 PM, Jason White wrote: > Chris Chen wrote: > > Brian, I would like to be one of the volunteers helping to report issues. > > That's great. We need more volunteers. > > For some FreeSWITCH users (of whom I am one), the user interface of Jira is > an > obstacle to reporting bugs via that mechanism, for accessibility reasons. > I've > had better luck with Bugzilla and other bug tracking systems, but Jira > really > seems to depend on Javascript for its fundamental operations, and that's a > real barrier for me at the moment. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/3824cae4/attachment-0001.html From brian at freeswitch.org Fri Jun 26 17:10:58 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 19:10:58 -0500 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090627000541.GB7509@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> Message-ID: what are the error messages? /b On Fri, Jun 26, 2009 at 7:05 PM, Jason White wrote: > Tamas wrote: > > Did you make bootstrap.sh and configure before compilation? > > Yes. This was a clean export from svn, built by running the Debian debuild > tool, as in > > svn export to a temporary directory, followed by debuild (after changing > the > version number to make the package version unique). > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/6b7326f1/attachment.html From jason at jasonjgw.net Fri Jun 26 17:31:40 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 10:31:40 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> Message-ID: <20090627003140.GA12881@jdc.jasonjgw.net> Brian West wrote: > what are the error messages? There aren't any. The build completes without error, but the module doesn't load due to the undefined symbols. From wiltingtree at gmail.com Fri Jun 26 19:10:10 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Fri, 26 Jun 2009 22:10:10 -0400 Subject: [Freeswitch-users] Version 1.0.4 not working with custom channel variables? Message-ID: Hi, I recently tried upgrading FreeSWITCH from 1.0.2 to 1.0.4pre8 on my server, but the application I wrote uses custom channel variables which I create using "setVariable" in Python. In version 1.0.2, I am able to retrieve that variable within the event xml (it prepends the variable name with variable_). But in version 1.0.4pre8, these variables are missing from the xml events. Could this be a bug? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/824b2ffc/attachment.html From vince.freeswitch at hightek.org Fri Jun 26 19:17:44 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Fri, 26 Jun 2009 21:17:44 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> Message-ID: <20090627021744.GA89233@quark.hightek.org> On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > > Can you post a bug to Jira.freeswitch.org with all these warnings, > even better with patches to fix it. OK. I think I have narrowed the problem down to 3 issues. 1. The build system is treating even a single warning as a critical error and aborting compilation in that directory. 2. Compilation continues into the next directory even though compilation of the previous directory was aborted, as you can see in the make output below where it went on to build "features". This can cause a chain reaction of other errors because stuff it expects to be there did not get built in previous stages. I suspect that is also the source of the 'symbolic link' error I was getting. I did not get that on this last compilation after fixing some of the warnings (see below). It can also make it possible to get to the end of the build and not know that stuff did not get compiled further back, leaving the package incomplete. 3. Lots of "return makes pointer from integer without a cast" warnings throughout the sofia-sip tree. This one, of course, is not the actual show stopper but is triggering the above problems and needs cleaned up none the less. ================= Making all in su LTCOMPILE su.lo LTCOMPILE su_errno.lo LTCOMPILE su_addrinfo.lo LTCOMPILE su_alloc.lo su_alloc.c: In function `sub_alloc': su_alloc.c:428: warning: return makes pointer from integer without a cast su_alloc.c:511: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_home_new': su_alloc.c:555: warning: return makes pointer from integer without a cast su_alloc.c:557: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_home_clone': su_alloc.c:730: warning: return makes pointer from integer without a cast su_alloc.c:732: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_realloc': su_alloc.c:1315: warning: return makes pointer from integer without a cast su_alloc.c:1319: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features ... ================= I confirmed the abort on warning issue (1) by fixing all warnings in su_alloc.c. As you can see below, it went past it just fine until it got a warning in su_sprintf.c. ================= Making all in su LTCOMPILE su.lo LTCOMPILE su_errno.lo LTCOMPILE su_addrinfo.lo LTCOMPILE su_alloc.lo LTCOMPILE su_alloc_lock.lo LTCOMPILE su_strdup.lo LTCOMPILE su_sprintf.lo su_sprintf.c: In function `su_vsprintf': su_sprintf.c:98: warning: return makes pointer from integer without a cast gmake[9]: *** [su_sprintf.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features ... ================= The issue below of saying it was successfull, when it was not, is apparently part of issue 2. It continued into the "build" directory after the previous errors. ================= Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 ================= I went ahead and established an account on jira.freeswitch.org. I see there are separate projects for "FreeSWITCH-Buildsystem" and "sofia-sip". I am guessing I should post two separate reports, one for issues 1 and 2 for the Buildsystem and one in sofia-sip for issue 3 since all the warnings seem to be from that code. I do not know the build system well enough to solve issues 1 and 2 at this time. However, I already have a patch I will provide for su_alloc.c and can continue working on resolving more of the casting warnings if somebody else can work on the build issue, which is the real show stopper. From vince.freeswitch at hightek.org Fri Jun 26 19:19:50 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Fri, 26 Jun 2009 21:19:50 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <9E2119E8-99A3-40B9-960F-539B881CF1EE@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <20090624031950.GD2623@hijacked.us> <20090625214910.GB45220@quark.hightek.org> <9E2119E8-99A3-40B9-960F-539B881CF1EE@jerris.com> Message-ID: <20090627021950.GB89233@quark.hightek.org> On Thu, Jun 25, 2009 at 06:06:04PM -0400, Michael Jerris wrote: > > > Is there a log file somewhere that contains the actual compile > > commands and error output so you can find out what happened when > > there is a error? Or perhaps a configuration to enable it to come > > out on the console? > > VERBOSE=1 gmake Thanks. That was helpfull. From brian at freeswitch.org Fri Jun 26 19:27:56 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 21:27:56 -0500 Subject: [Freeswitch-users] Version 1.0.4 not working with custom channel variables? In-Reply-To: References: Message-ID: <3372AF70-46CB-4B71-8455-B8E1719D22D0@freeswitch.org> Execute the app verbose_events /b On Jun 26, 2009, at 9:10 PM, Adam Wilt wrote: > Hi, I recently tried upgrading FreeSWITCH from 1.0.2 to 1.0.4pre8 on > my server, but the application I wrote uses custom channel variables > which I create using "setVariable" in Python. In version 1.0.2, I am > able to retrieve that variable within the event xml (it prepends the > variable name with variable_). But in version 1.0.4pre8, these > variables are missing from the xml events. Could this be a bug? > Thanks! > > From jason at jasonjgw.net Fri Jun 26 19:31:02 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 12:31:02 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090627003140.GA12881@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> <20090627003140.GA12881@jdc.jasonjgw.net> Message-ID: <20090627023102.GA21650@jdc.jasonjgw.net> Sorry - I misread the question. The error is: freeswitch at default> load mod_portaudio -ERR [module load file routine returned an error] 2009-06-27 12:30:00.740316 [CRIT] switch_loadable_module.c:871 Error Loading mod ule /opt/freeswitch/mod/mod_portaudio.so **/opt/freeswitch/mod/mod_portaudio.so: undefined symbol: snd_config** freeswitch at default> There is no error during the building. From lon at kickasspixels.com Fri Jun 26 19:50:21 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 26 Jun 2009 19:50:21 -0700 Subject: [Freeswitch-users] Event socket chunks don't always end with 2 line breaks? Message-ID: <5d3e0dc60906261950l60af9635r964049ed3ac976d8@mail.gmail.com> In the latest SVN this seem to be true for CHANNEL_HANGUP, which appends the following after the normal line breaks. "Disconnected, goodbye.\n See you at ClueCon! http://www.cluecon.com/" Perhaps it can be changed to conform the chunked data structure? variable_disconnect_message: Disconnected, goodbye. See you at ClueCon! http://www.cluecon.com Just a thought. Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/01b9c7a5/attachment.html From brian at freeswitch.org Fri Jun 26 19:54:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 21:54:19 -0500 Subject: [Freeswitch-users] Event socket chunks don't always end with 2 line breaks? In-Reply-To: <5d3e0dc60906261950l60af9635r964049ed3ac976d8@mail.gmail.com> References: <5d3e0dc60906261950l60af9635r964049ed3ac976d8@mail.gmail.com> Message-ID: <2D44EBE1-115A-4305-AF37-BDAF85084271@freeswitch.org> The protocol has a content-length header.. you read exactly that many bytes and you won't be confused by this disconnect message. /b On Jun 26, 2009, at 9:50 PM, Lon Baker wrote: > In the latest SVN this seem to be true for CHANNEL_HANGUP, which > appends the following after the normal line breaks. > > "Disconnected, goodbye.\n > See you at ClueCon! http://www.cluecon.com/" > > Perhaps it can be changed to conform the chunked data structure? > > variable_disconnect_message: Disconnected, goodbye. See you at > ClueCon! http://www.cluecon.com > > Just a thought. > > Lon > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/0929f5b3/attachment-0001.html From lon at kickasspixels.com Fri Jun 26 20:06:54 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 26 Jun 2009 20:06:54 -0700 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 270 In-Reply-To: References: Message-ID: <5d3e0dc60906262006v7aad7b5dx549d6739898f48a4@mail.gmail.com> Brian, Thanks! I was just about to post a "never mind" after further reading source. Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/53b22cd0/attachment.html From mike at jerris.com Fri Jun 26 20:19:02 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 23:19:02 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090627021744.GA89233@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> <20090627021744.GA89233@quark.hightek.org> Message-ID: <0CFF5BFD-E500-4E8A-9802-F6BA0519B522@jerris.com> On Jun 26, 2009, at 10:17 PM, Vincent Stemen wrote: > On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: >> >> Can you post a bug to Jira.freeswitch.org with all these warnings, >> even better with patches to fix it. > > OK. I think I have narrowed the problem down to 3 issues. > > 1. The build system is treating even a single warning as a critical > error > and aborting compilation in that directory. As intended. > 2. Compilation continues into the next directory even though > compilation of > the previous directory was aborted, as you can see in the make > output below > where it went on to build "features". This can cause a chain > reaction of > other errors because stuff it expects to be there did not get > built in > previous stages. I suspect that is also the source of the > 'symbolic link' > error I was getting. I did not get that on this last compilation > after > fixing some of the warnings (see below). It can also make it > possible to > get to the end of the build and not know that stuff did not get > compiled > further back, leaving the package incomplete. > I've looked for this one and have not been able to nail it down. I must be missing an || exit somewhere in the module makefiles? Everytime I go to reproduce this issue I can't reproduce it. > 3. Lots of "return makes pointer from integer without a cast" > warnings > throughout the sofia-sip tree. This one, of course, is not the > actual show > stopper but is triggering the above problems and needs cleaned up > none the > less. I am sure these are trivial enough to fix but am a bit puzzled why I don't see them on. Any other platform. What version of gcc is this? Does dragonfly patch gcc to report more warnings than other platforms? > > > ================= > > Making all in su > LTCOMPILE su.lo > LTCOMPILE su_errno.lo > LTCOMPILE su_addrinfo.lo > LTCOMPILE su_alloc.lo > su_alloc.c: In function `sub_alloc': > su_alloc.c:428: warning: return makes pointer from integer without a > cast > su_alloc.c:511: warning: return makes pointer from integer without a > cast > su_alloc.c: In function `su_home_new': > su_alloc.c:555: warning: return makes pointer from integer without a > cast > su_alloc.c:557: warning: return makes pointer from integer without a > cast > su_alloc.c: In function `su_home_clone': > su_alloc.c:730: warning: return makes pointer from integer without a > cast > su_alloc.c:732: warning: return makes pointer from integer without a > cast > su_alloc.c: In function `su_realloc': > su_alloc.c:1315: warning: return makes pointer from integer without > a cast > su_alloc.c:1319: warning: return makes pointer from integer without > a cast > su_alloc.c: In function `su_salloc': > su_alloc.c:1518: warning: return makes pointer from integer without > a cast > gmake[9]: *** [su_alloc.lo] Error 1 > gmake[8]: *** [all] Error 2 > Making all in features > ... > > ================= > I confirmed the abort on warning issue (1) by fixing all warnings in > su_alloc.c. As you can see below, it went past it just fine until > it got > a warning in su_sprintf.c. > ================= > > Making all in su > LTCOMPILE su.lo > LTCOMPILE su_errno.lo > LTCOMPILE su_addrinfo.lo > LTCOMPILE su_alloc.lo > LTCOMPILE su_alloc_lock.lo > LTCOMPILE su_strdup.lo > LTCOMPILE su_sprintf.lo > su_sprintf.c: In function `su_vsprintf': > su_sprintf.c:98: warning: return makes pointer from integer without > a cast > gmake[9]: *** [su_sprintf.lo] Error 1 > gmake[8]: *** [all] Error 2 > Making all in features > ... > > ================= > The issue below of saying it was successfull, when it was not, is > apparently > part of issue 2. It continued into the "build" directory after the > previous > errors. > ================= > > Making all in packages > gmake[6]: *** [all-recursive] Error 1 > gmake[5]: *** [all] Error 2 > gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- > ua.la] Error 2 > gmake[3]: *** [mod_sofia-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + gmake install + > +----------------------------------------------+ > gmake[1]: *** [all-recursive] Error 1 > gmake: *** [all] Error 2 > > ================= > > I went ahead and established an account on jira.freeswitch.org. I > see there > are separate projects for "FreeSWITCH-Buildsystem" and "sofia-sip". I > am guessing I should post two separate reports, one for issues 1 and > 2 for the > Buildsystem and one in sofia-sip for issue 3 since all the warnings > seem to be > from that code. > > I do not know the build system well enough to solve issues 1 and 2 > at this > time. However, I already have a patch I will provide for su_alloc.c > and can > continue working on resolving more of the casting warnings if > somebody else can > work on the build issue, which is the real show stopper. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Fri Jun 26 20:30:10 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 26 Jun 2009 20:30:10 -0700 Subject: [Freeswitch-users] att_xfer w/uuid Message-ID: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> I'm trying to use xml_rpc to initiate an att_xfer on channel A (which is already bridged to channel B), but I'm running into some issues. I know the uuid's from both channel A and B, but the documentation I found on att_xfer only seems to indicate a way to do this from DMTF presses occurring on channel A. my idea was to use xml_rpc to execute a lua script which would take a uuid as an argv and bind to the session with freeswitch.session(uuid). I tried this, but the audio breaks up with the session that the lua script binded too. Does anyone have any recommendations on how I might accomplish an assisted transfer w/o DTMF presses and bind_meta_app knowing only a uuid? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/2e47aa3e/attachment.html From brian at freeswitch.org Fri Jun 26 21:08:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 23:08:16 -0500 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> Message-ID: <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> Not sure what you want to do is doable via XML RPC. That app is to be run on an existing session. The other solution is to take both legs and park them.. Then execute bridge on one leg to the target transfer person. Once that call is up.. you can them park both of those.. uuid_bridge the two you wish to complete then hang up on the third one. I think if you just uuid_bridge the two you want in the end the third one will just hangup. /b On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: > I'm trying to use xml_rpc to initiate an att_xfer on channel A > (which is already bridged to channel B), but I'm running into some > issues. > > I know the uuid's from both channel A and B, but the documentation I > found on att_xfer only seems to indicate a way to do this from DMTF > presses occurring on channel A. > > my idea was to use xml_rpc to execute a lua script which would take > a uuid as an argv and bind to the session with > freeswitch.session(uuid). > > I tried this, but the audio breaks up with the session that the lua > script binded too. Does anyone have any recommendations on how I > might accomplish an assisted transfer w/o DTMF presses and > bind_meta_app knowing only a uuid? > > Thanks. > > --matt From mattdfong at gmail.com Fri Jun 26 21:30:43 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 26 Jun 2009 21:30:43 -0700 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> Message-ID: <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> can you 3 way with uuid_bridge? --matt On Fri, Jun 26, 2009 at 9:08 PM, Brian West wrote: > Not sure what you want to do is doable via XML RPC. That app is to be > run on an existing session. The other solution is to take both legs > and park them.. Then execute bridge on one leg to the target transfer > person. Once that call is up.. you can them park both of those.. > uuid_bridge the two you wish to complete then hang up on the third > one. I think if you just uuid_bridge the two you want in the end the > third one will just hangup. > > /b > > On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: > > > I'm trying to use xml_rpc to initiate an att_xfer on channel A > > (which is already bridged to channel B), but I'm running into some > > issues. > > > > I know the uuid's from both channel A and B, but the documentation I > > found on att_xfer only seems to indicate a way to do this from DMTF > > presses occurring on channel A. > > > > my idea was to use xml_rpc to execute a lua script which would take > > a uuid as an argv and bind to the session with > > freeswitch.session(uuid). > > > > I tried this, but the audio breaks up with the session that the lua > > script binded too. Does anyone have any recommendations on how I > > might accomplish an assisted transfer w/o DTMF presses and > > bind_meta_app knowing only a uuid? > > > > Thanks. > > > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/065b047d/attachment.html From dome at tel.co.th Fri Jun 26 23:45:36 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 27 Jun 2009 13:45:36 +0700 Subject: [Freeswitch-users] How to cancel session in Javascript Message-ID: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Dear All, I try s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); if (s.ready()){ s.setVariable("nibble_rate", "2.5"); s.setVariable("nibble_account", "0838833133"); s.execute("nibblebill", "heartbeat 5"); bridge(session,s); }; my question is 1. How to cancel create s session (by dtmf ) like a * in bridge app 2. when i hangup before s session ready is posible to cancel ? Best Regards. Dome C. From jason at jasonjgw.net Sat Jun 27 04:00:35 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 21:00:35 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090627023102.GA21650@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> <20090627003140.GA12881@jdc.jasonjgw.net> <20090627023102.GA21650@jdc.jasonjgw.net> Message-ID: <20090627110035.GA10140@jdc.jasonjgw.net> As a further note on this subject, temporarily downgrading to libtool 1.5.26 and rebuilding FreeSWITCH gave me a working mod_portaudio.so module. Obviously this doesn't solve the problem, but it does prove that, as suspected, the migration to libtool 2.2.6a was the cause. Any suggestions on how to track down the build system bug would be welcome. From msc at freeswitch.org Sat Jun 27 10:09:32 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 27 Jun 2009 10:09:32 -0700 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> Message-ID: <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> Couldn't you just throw all the calls into a conference at this point? -MC Sent from my iPhone On Jun 26, 2009, at 9:30 PM, Matthew Fong wrote: > can you 3 way with uuid_bridge? > > --matt > > On Fri, Jun 26, 2009 at 9:08 PM, Brian West > wrote: > Not sure what you want to do is doable via XML RPC. That app is to be > run on an existing session. The other solution is to take both legs > and park them.. Then execute bridge on one leg to the target transfer > person. Once that call is up.. you can them park both of those.. > uuid_bridge the two you wish to complete then hang up on the third > one. I think if you just uuid_bridge the two you want in the end the > third one will just hangup. > > /b > > On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: > > > I'm trying to use xml_rpc to initiate an att_xfer on channel A > > (which is already bridged to channel B), but I'm running into some > > issues. > > > > I know the uuid's from both channel A and B, but the documentation I > > found on att_xfer only seems to indicate a way to do this from DMTF > > presses occurring on channel A. > > > > my idea was to use xml_rpc to execute a lua script which would take > > a uuid as an argv and bind to the session with > > freeswitch.session(uuid). > > > > I tried this, but the audio breaks up with the session that the lua > > script binded too. Does anyone have any recommendations on how I > > might accomplish an assisted transfer w/o DTMF presses and > > bind_meta_app knowing only a uuid? > > > > Thanks. > > > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/95b6986e/attachment-0001.html From brian at freeswitch.org Sat Jun 27 16:05:58 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 27 Jun 2009 18:05:58 -0500 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> Message-ID: <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> my thinking exactly. /b On Jun 27, 2009, at 12:09 PM, Michael S Collins wrote: > Couldn't you just throw all the calls into a conference at this point? > -MC > > Sent from my iPhone From Mailings at kh-dev.de Sat Jun 27 16:47:08 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 28 Jun 2009 01:47:08 +0200 Subject: [Freeswitch-users] Result of an application... Message-ID: Hi, maybe a stupid question, but how can I find out the result of an application? If I do (in perl) $session->execute("bridge", "user/${DestNr}\@$VARS{domain}"); How do I know if this was successful or if the user was busy or if the phone doesn't exist? Is there any status variable for the result of an execute (or even any other command)? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/eba58ba1/attachment.html From mattdfong at gmail.com Sat Jun 27 19:18:48 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Sat, 27 Jun 2009 19:18:48 -0700 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> Message-ID: <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> there's a reference on the wiki to a three_way dial plan command. What does that do? What's the best way to put 2 bridged callers into a new conference? Must I park both uuid's first, and then transfer both to an extension that will add them to a new conference? Is there a way to do this without any break in the audio? Thanks... --matt On Sat, Jun 27, 2009 at 4:05 PM, Brian West wrote: > my thinking exactly. > > /b > > On Jun 27, 2009, at 12:09 PM, Michael S Collins wrote: > > > Couldn't you just throw all the calls into a conference at this point? > > -MC > > > > Sent from my iPhone > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/1b1248c0/attachment.html From jason at jasonjgw.net Sat Jun 27 19:32:37 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 28 Jun 2009 12:32:37 +1000 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> Message-ID: <20090628023237.GA7470@jdc.jasonjgw.net> Matthew Fong wrote: > What's the best way to put 2 bridged callers into a new conference? Must I > park both uuid's first, and then transfer both to an extension that will add > them to a new conference? No, it's uuid_transfer with the -both option to transfer both legs to the conference extension. From brian at freeswitch.org Sat Jun 27 19:39:27 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 27 Jun 2009 21:39:27 -0500 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> Message-ID: <0EEF60D0-2FE1-424F-B1B5-5234C8EDC04F@freeswitch.org> yes and in your case you can forget it. With what you wanna do its not possible to use that. /b On Jun 27, 2009, at 9:18 PM, Matthew Fong wrote: > there's a reference on the wiki to a three_way dial plan command. > What does that do? > > What's the best way to put 2 bridged callers into a new conference? > Must I park both uuid's first, and then transfer both to an > extension that will add them to a new conference? Is there a way to > do this without any break in the audio? Thanks... > > --matt From vince.freeswitch at hightek.org Sat Jun 27 22:17:06 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Sun, 28 Jun 2009 00:17:06 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <0CFF5BFD-E500-4E8A-9802-F6BA0519B522@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> <20090627021744.GA89233@quark.hightek.org> <0CFF5BFD-E500-4E8A-9802-F6BA0519B522@jerris.com> Message-ID: <20090628051706.GA13252@quark.hightek.org> On Fri, Jun 26, 2009 at 11:19:02PM -0400, Michael Jerris wrote: > > > On Jun 26, 2009, at 10:17 PM, Vincent Stemen > wrote: > > > On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > >> > >> Can you post a bug to Jira.freeswitch.org with all these warnings, > >> even better with patches to fix it. > > > > OK. I think I have narrowed the problem down to 3 issues. > > > > 1. The build system is treating even a single warning as a critical > > error > > and aborting compilation in that directory. > > As intended. Interesting. I don't think I have seen any other projects force the build to fail on warnings. Although, I think that is better than allowing warnings to accumulate and never get cleaned up like I see on a lot of the projects out there. > > 2. Compilation continues into the next directory even though > > compilation of > > the previous directory was aborted, as you can see in the make > > output below > > where it went on to build "features". This can cause a chain > > reaction of > > other errors because stuff it expects to be there did not get > > built in > > previous stages. I suspect that is also the source of the > > 'symbolic link' > > error I was getting. I did not get that on this last compilation > > after > > fixing some of the warnings (see below). It can also make it > > possible to > > get to the end of the build and not know that stuff did not get > > compiled > > further back, leaving the package incomplete. > > > > I've looked for this one and have not been able to nail it down. I > must be missing an || exit somewhere in the module makefiles? > Everytime I go to reproduce this issue I can't reproduce it. Since you are aware of it and working on it, do you still want me to take the time to create a problem report on jira for this? If so, do you want to add a platform option for Dragonfly BSD first? Or should I just select FreeBSD? > > 3. Lots of "return makes pointer from integer without a cast" > > warnings > > throughout the sofia-sip tree. This one, of course, is not the > > actual show > > stopper but is triggering the above problems and needs cleaned up > > none the > > less. > > I am sure these are trivial enough to fix but am a bit puzzled why I > don't see them on. Any other platform. What version of gcc is this? > Does dragonfly patch gcc to report more warnings than other platforms? Hmm.. It could be that you guys are all running gcc-4 (?). I am working on a bit older installation that is running gcc 3.4.6. Although, I was told on the dragonfly irc that gcc-4 generates more warnings by default than gcc-3. So I don't know. They said they did not know of any special gcc configurations or patches for gcc on Dragonfly. There was one thing unanimous though. Everybody I spoke with on the #c and the #dragonflybsd irc did not like the way the lines are coded that are generating most of the errors :-). e.g. return (void)(errno = EINVAL), NULL; I went ahead and posted a bug report on jira under the sofia-sip project, with a patch that fixes all the warnings for the first file (su_alloc.c). More patches will follow. I went ahead and selected FreeBSD as the platform. I thought I would point that out in case you guys want to add Dragonfly BSD and change the platform on this issue report. From nicolas at medularis.com Sun Jun 28 09:16:49 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Sun, 28 Jun 2009 12:16:49 -0400 Subject: [Freeswitch-users] hangup_cause NONE vs. NORMAL_CLEARING Message-ID: <1b46b4e80906280916g5fb065e9w65183f9117609f68@mail.gmail.com> I have a small JS script that makes a call, plays a sound file and then hangs up. For each call it makes, I log the hangup_cause variable on the CHANNEL_HANGUP_COMPLETE event. Most of the time, when calls are successful, I get a NORMAL_CLEARING cause, but sometimes I'll get a NONE cause. I wanted to know what the difference between these two is, because there is no reference to NONE in the wiki (http://wiki.freeswitch.org/wiki/Hangup_causes ). Thanks, Nicol?s -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/9b5cc05b/attachment.html From nik.middleton at noblesolutions.co.uk Sun Jun 28 11:49:18 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 19:49:18 +0100 Subject: [Freeswitch-users] Confused with event content lengths Message-ID: Hi Guys, I'm trying to parse events in C++ for an outbound socket. The docs are a little contradictory, so I wonder if someone could help me out. As I understand it and event is terminated with double LF's (\n\n) However if there is a Content-Length header the wiki very confusingly says 'Content-Length is the length of the event beginning AFTER the very next LF only line ("\n") and inclusive the trailing LF/LF pair ("\n\n")' BUT the example says it's after the \n\n in the header!! Which is it? In addition, it also looks like the event body is also terminated by a \n\n. If this is the case, why do I care about content length value, can't I simply read until I get the termination sequence? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/eaba5298/attachment-0001.html From jmesquita at gmail.com Sun Jun 28 11:57:21 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 28 Jun 2009 15:57:21 -0300 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: References: Message-ID: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> If I am not mistaken, you are always safe reading the amount data expressed on Content-Length since it is calculated based on the total message length before it is sent out of FS. >From a protocol point of view, it would indeed be much better to rely on something such as Content-Length then \n\n termination string. As I get to know more and more the core developers, I doubt they would rely on the latter. Hope it helps... jmesquita On Sun, Jun 28, 2009 at 3:49 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?m trying to parse events in C++ for an outbound socket. The docs are a > little contradictory, so I wonder if someone could help me out. > > > > As I understand it and event is terminated with double LF?s (\n\n) However > if there is a Content-Length header the wiki very confusingly says > > > > ?Content-Length is the length of the event beginning *AFTER* the very next > LF only line ("\n") and *inclusive* the trailing LF/LF pair ("\n\n")? > > > > BUT the example says it?s after the \n\n in the header!! Which is it? > > > > In addition, it also looks like the event body is also terminated by a > \n\n. If this is the case, why do I care about content length value, can?t > I simply read until I get the termination sequence? > > > > Regards, > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/cb93a7dc/attachment.html From brian at freeswitch.org Sun Jun 28 12:23:28 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 28 Jun 2009 14:23:28 -0500 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> References: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> Message-ID: <15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> Yes it says 264 bytes read exactly 264 bytes or die trying. /b On Jun 28, 2009, at 1:57 PM, Jo?o Mesquita wrote: > If I am not mistaken, you are always safe reading the amount data > expressed on Content-Length since it is calculated based on the > total message length before it is sent out of FS. > > From a protocol point of view, it would indeed be much better to > rely on something such as Content-Length then \n\n termination > string. As I get to know more and more the core developers, I doubt > they would rely on the latter. > > Hope it helps... > > jmesquita From danishmoosa at gmail.com Sun Jun 28 04:30:44 2009 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Sun, 28 Jun 2009 17:30:44 +0600 Subject: [Freeswitch-users] bypass media mode is stateful? Message-ID: Hi, Although its obvious but just asking for confirmation :) Is bypass media mode in FS is fairly stateful. If it in huge network so its cdr(billing) is relaiable ? -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/a15b03d7/attachment.html From brian at freeswitch.org Sun Jun 28 12:30:07 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 28 Jun 2009 14:30:07 -0500 Subject: [Freeswitch-users] bypass media mode is stateful? In-Reply-To: References: Message-ID: <77E11941-BD88-4CA9-83A2-D7C77376A190@freeswitch.org> Yes CDR is still valid.. the media is just bypassed... please review the FAQ. /b On Jun 28, 2009, at 6:30 AM, Muhammad Danish Moosa wrote: > Hi, > > Although its obvious but just asking for confirmation :) > > Is bypass media mode in FS is fairly stateful. If it in huge network > so its cdr(billing) is relaiable ? > From nik.middleton at noblesolutions.co.uk Sun Jun 28 12:38:01 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 20:38:01 +0100 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: <15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> References: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> <15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> Message-ID: But from where? After the double LF of the header as one part of the wiki says or after the line containing the content-length that another part of the wiki says? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 28 June 2009 20:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Confused with event content lengths Yes it says 264 bytes read exactly 264 bytes or die trying. /b On Jun 28, 2009, at 1:57 PM, Jo?o Mesquita wrote: > If I am not mistaken, you are always safe reading the amount data > expressed on Content-Length since it is calculated based on the > total message length before it is sent out of FS. > > From a protocol point of view, it would indeed be much better to > rely on something such as Content-Length then \n\n termination > string. As I get to know more and more the core developers, I doubt > they would rely on the latter. > > Hope it helps... > > jmesquita _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Jun 28 15:40:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 23:40:20 +0100 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: References: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com><15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> Message-ID: OK, finally figured it out. Have updated the Wiki to remove ambiguity and posted some SUDO code Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 28 June 2009 20:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Confused with event content lengths But from where? After the double LF of the header as one part of the wiki says or after the line containing the content-length that another part of the wiki says? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 28 June 2009 20:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Confused with event content lengths Yes it says 264 bytes read exactly 264 bytes or die trying. /b On Jun 28, 2009, at 1:57 PM, Jo?o Mesquita wrote: > If I am not mistaken, you are always safe reading the amount data > expressed on Content-Length since it is calculated based on the > total message length before it is sent out of FS. > > From a protocol point of view, it would indeed be much better to > rely on something such as Content-Length then \n\n termination > string. As I get to know more and more the core developers, I doubt > they would rely on the latter. > > Hope it helps... > > jmesquita _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Jun 28 15:55:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 23:55:14 +0100 Subject: [Freeswitch-users] Myevents in outbound socket Message-ID: Hi Guys, I've almost got my c++ outbound socket control prog running, however even though the filter works, it would be truly great to just subscribe to myevents as even with the filter in place I get lots of channel Execute and complete events which I don't really need. Problem is, is that mod_VMD isn't included in those events, even though it is channel specific. Is there any chance that this will be included? If not, can someone point me to where myevents is defined and I'll have a go at it myself. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/9ac0b4db/attachment.html From brian at freeswitch.org Sun Jun 28 15:59:56 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 28 Jun 2009 17:59:56 -0500 Subject: [Freeswitch-users] Myevents in outbound socket In-Reply-To: References: Message-ID: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> Are you using ESL? /b On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote: > Hi Guys, > > I?ve almost got my c++ outbound socket control prog running, however > even though the filter works, it would be truly great to just > subscribe to myevents as even with the filter in place I get lots of > channel Execute and complete events which I don?t really need. > Problem is, is that mod_VMD isn?t included in those events, even > though it is channel specific. Is there any chance that this will > be included? If not, can someone point me to where myevents is > defined and I?ll have a go at it myself. > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/2f02ac81/attachment-0001.html From nik.middleton at noblesolutions.co.uk Sun Jun 28 16:14:21 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 00:14:21 +0100 Subject: [Freeswitch-users] Myevents in outbound socket In-Reply-To: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> References: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> Message-ID: Nope. Can't find much on the Wiki on how to interface with ESL using C++. I want to control the outbound socket from a windows 2003 server only because that's what I'm familiar with. Is there some portable C++ or C code? ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 29 June 2009 00:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Myevents in outbound socket Are you using ESL? /b On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote: Hi Guys, I've almost got my c++ outbound socket control prog running, however even though the filter works, it would be truly great to just subscribe to myevents as even with the filter in place I get lots of channel Execute and complete events which I don't really need. Problem is, is that mod_VMD isn't included in those events, even though it is channel specific. Is there any chance that this will be included? If not, can someone point me to where myevents is defined and I'll have a go at it myself. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/5ff0a126/attachment.html From jmesquita at gmail.com Sun Jun 28 16:46:22 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 28 Jun 2009 20:46:22 -0300 Subject: [Freeswitch-users] Myevents in outbound socket In-Reply-To: References: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> Message-ID: <5a8712120906281646ma945857h942f72fff4c25ac2@mail.gmail.com> You should definitely look at ESL, dude. Take a look at ${SVNROOT}/libs/esl/. There is a esl_oop inside that might give you a go. Beware that this is only an interface for SWIG, but might be useful to you if you extend it. Later, jmesquita On Sun, Jun 28, 2009 at 8:14 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Nope. > > > > Can?t find much on the Wiki on how to interface with ESL using C++. I want > to control the outbound socket from a windows 2003 server only because > that?s what I?m familiar with. Is there some portable C++ or C code? > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* 29 June 2009 00:00 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Myevents in outbound socket > > > > Are you using ESL? > > > > /b > > > > On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote: > > > > Hi Guys, > > > > I?ve almost got my c++ outbound socket control prog running, however even > though the filter works, it would be truly great to just subscribe to > myevents as even with the filter in place I get lots of channel Execute and > complete events which I don?t really need. Problem is, is that mod_VMD > isn?t included in those events, even though it is channel specific. Is > there any chance that this will be included? If not, can someone point me > to where myevents is defined and I?ll have a go at it myself. > > > > Regards, > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/d2cdbbfc/attachment.html From yudha2008 at gmail.com Sun Jun 28 21:53:15 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 29 Jun 2009 10:23:15 +0530 Subject: [Freeswitch-users] javascript FIFO (First In First Out) Message-ID: *Hi, I have configured inbound through JavaScript it is working well. Through dialplan i have configured FIFO it is also working fine but i want to configure FIFO (First In First Out) through JavaScript. Is there any link or examples to configure FIFO through JavaScript which will assist me to resolve this problem. can any one assist me to do this above process. Thanks in advance. Thanks with Regards, N.Baskar. * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/49b52cdb/attachment.html From mprabhuram at gmail.com Mon Jun 29 00:23:58 2009 From: mprabhuram at gmail.com (Prabhuram Mohan) Date: Mon, 29 Jun 2009 00:23:58 -0700 Subject: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC Message-ID: Hi All, Abode AIR based on flash player can be used to connect to freeswitch and issue commands through XMLRPC. I read the internet to do this plumbing and also got help through generous fellow developers from AIR & freeswitch community. Now that it is successfully done, here is the complete code for the benefit of people who are following suite. Comments are welcome!! Code available here - http://neoalchemist.tumblr.com/post/132134683 Thanks Prabhu From shaheryarkh at googlemail.com Mon Jun 29 00:42:27 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 29 Jun 2009 08:42:27 +0100 Subject: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC In-Reply-To: References: Message-ID: Good job man! This is really useful. Thank you. On Mon, Jun 29, 2009 at 8:23 AM, Prabhuram Mohan wrote: > Hi All, > > Abode AIR based on flash player can be used to connect to freeswitch > and issue commands through XMLRPC. I read the internet to do this > plumbing and also got help through generous fellow developers from AIR > & freeswitch community. Now that it is successfully done, here is the > complete code for the benefit of people who are following suite. > Comments are welcome!! > > Code available here - http://neoalchemist.tumblr.com/post/132134683 > > Thanks > Prabhu > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/80018d38/attachment.html From jingwei.yang at gmail.com Mon Jun 29 02:25:09 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 29 Jun 2009 17:25:09 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> Message-ID: <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> Hi Chris, any thoughts? Thanks, -Jingwei On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang wrote: > Hi Chris, here's the one that confuses me. As far as I understand, the > extension 888 defined in public.xml is for picking up incoming calls. It > should have no influence on outgoing calls, right? If not, what is to write > to fit my case? (originate dingaling/gmail.com/userAAA at gmail.com&bridge(dingaling/ > gmail.com/userBBB at gmail.com), both userAAA and userBBB can be internal or > external). > > Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm > not quite sure what to include. So I make it very simple. > > > > > > > > Here are three relative parameters in client.xml: > > > > > > Still, I got no echo for internal Ip calls. Please let me know where goes > wrong. > > Thanks, > -Jingwei > > On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen wrote: > >> I guess you have the problem here, >> in client.xml you have >> >> >> but you only define extension 888 in default context, >> that's why nobody can reach you from public. >> >> under /usr/local/freeswitch/conf/dialplan >> >> define extension 888 in public.xml to the proper extension you expect, and >> check the console log from fs_cli when you do gtalk calling to your gmail >> client, you will find out the solution to your issue. >> >> chris >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/da9fddaf/attachment-0001.html From darklion11 at yahoo.com Mon Jun 29 03:44:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 03:44:41 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? Message-ID: <24251951.post@talk.nabble.com> Hi, Is there any available license G729 for freeswitch? I need it to match G729 of Asterisks? If any please help me and instructions how to install this thing. Thanks -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24251951.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From chris.chen2004 at gmail.com Mon Jun 29 03:46:13 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Mon, 29 Jun 2009 06:46:13 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> Message-ID: <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> Jingwei, I don't know if you have the 888 defined in default.xml? also you have to define $${domain}. please do " dl_debug on" from fs_cli, and watch the console logs and see what's going on when you try calling from external. Most likely your dialplan is not correctly defined. Chris On Mon, Jun 29, 2009 at 5:25 AM, Jingwei Yang wrote: > Hi Chris, any thoughts? > > Thanks, > -Jingwei > > > On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang wrote: > >> Hi Chris, here's the one that confuses me. As far as I understand, the >> extension 888 defined in public.xml is for picking up incoming calls. It >> should have no influence on outgoing calls, right? If not, what is to write >> to fit my case? (originate dingaling/gmail.com/userAAA at gmail.com&bridge(dingaling/ >> gmail.com/userBBB at gmail.com), both userAAA and userBBB can be internal or >> external). >> >> Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm >> not quite sure what to include. So I make it very simple. >> >> >> >> >> >> >> >> Here are three relative parameters in client.xml: >> >> >> >> >> >> Still, I got no echo for internal Ip calls. Please let me know where goes >> wrong. >> >> Thanks, >> -Jingwei >> >> On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen wrote: >> >>> I guess you have the problem here, >>> in client.xml you have >>> >>> >>> but you only define extension 888 in default context, >>> that's why nobody can reach you from public. >>> >>> under /usr/local/freeswitch/conf/dialplan >>> >>> define extension 888 in public.xml to the proper extension you expect, >>> and check the console log from fs_cli when you do gtalk calling to your >>> gmail client, you will find out the solution to your issue. >>> >>> chris >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/eb19bc01/attachment.html From jason at jasonjgw.net Mon Jun 29 03:51:13 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 29 Jun 2009 20:51:13 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24251951.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> Message-ID: <20090629105113.GA4756@jdc.jasonjgw.net> Edmar Cruz wrote: > > Is there any available license G729 for freeswitch? Yes. It was announced here a few days ago - see the list archives. From darklion11 at yahoo.com Mon Jun 29 04:01:02 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 04:01:02 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <20090629105113.GA4756@jdc.jasonjgw.net> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> Message-ID: <24252099.post@talk.nabble.com> Yup. I got this mid_dahdi_codec but instructions how to install it? Jason White-14 wrote: > > Edmar Cruz wrote: >> >> Is there any available license G729 for freeswitch? > > Yes. It was announced here a few days ago - see the list archives. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24252099.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From odermann at googlemail.com Mon Jun 29 05:30:11 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 29 Jun 2009 14:30:11 +0200 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> Message-ID: <5e414ed0906290530q2fdb0643k34d9f883dc52deec@mail.gmail.com> thanks for your answers. i did not know, that muxing is cpu intensive. i just thought, that it would not matter, if one is muxing 50/50 or 30/70. for playing back a soundfile, while one can hear the other end it seems, that muxing is required. so the level of muxing should not make a difference!? anyway, if there is no other/better way, we have to do it with sox. no, we are not using stereo-files. kind regards dennis From javieraristizabal at gmail.com Mon Jun 29 06:47:05 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Mon, 29 Jun 2009 08:47:05 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24252099.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> Message-ID: mid_dahdi_codec is the source to build the TC400B card. On Mon, Jun 29, 2009 at 6:01 AM, Edmar Cruz wrote: > > Yup. I got this mid_dahdi_codec but instructions how to install it? > > Jason White-14 wrote: >> >> Edmar Cruz wrote: >>> >>> ? Is there any available license G729 for freeswitch? >> >> Yes. It was announced here a few days ago - see the list archives. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24252099.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Jun 29 06:51:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 08:51:42 -0500 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <5e414ed0906290530q2fdb0643k34d9f883dc52deec@mail.gmail.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> <5e414ed0906290530q2fdb0643k34d9f883dc52deec@mail.gmail.com> Message-ID: <39237CF4-51E2-4044-8302-24DF77DEBF50@freeswitch.org> Yes muxing like this will cause the volume to go up a little bit depending on the source input file. /b On Jun 29, 2009, at 7:30 AM, Dennis wrote: > thanks for your answers. i did not know, that muxing is cpu intensive. > i just thought, that it would not matter, if one is muxing 50/50 or > 30/70. for playing back a soundfile, while one can hear the other end > it seems, that muxing is required. so the level of muxing should not > make a difference!? > > anyway, if there is no other/better way, we have to do it with sox. > > no, we are not using stereo-files. > > > kind regards > dennis From mike at jerris.com Mon Jun 29 08:06:50 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:06:50 -0400 Subject: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC In-Reply-To: References: Message-ID: <481FFA3C-C0F8-4B32-A4DE-D53498D2B547@jerris.com> If you are interested, we can create a contrib directory for you in the codebase and you can commit it right in tree. If interested, please log on to irc and we can coordinate setting up an account for you. Mike On Jun 29, 2009, at 3:23 AM, Prabhuram Mohan wrote: > Hi All, > > Abode AIR based on flash player can be used to connect to freeswitch > and issue commands through XMLRPC. I read the internet to do this > plumbing and also got help through generous fellow developers from AIR > & freeswitch community. Now that it is successfully done, here is the > complete code for the benefit of people who are following suite. > Comments are welcome!! > > Code available here - http://neoalchemist.tumblr.com/post/132134683 > > Thanks > Prabhu > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shiyanov at gmail.com Mon Jun 29 08:07:57 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 29 Jun 2009 19:07:57 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: I've tried with the snapshot (06.26.2009) - and situation had become even worse - now both agents hear nothing.. Maybe problem is in my sip_profiles? Here they are: http://pastebin.freeswitch.org/pastebin.php?dl=9510 http://pastebin.freeswitch.org/pastebin.php?dl=9511 On Fri, Jun 26, 2009 at 10:03 PM, Brian West wrote: > Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk > also... due to the lines below. > > /b > > On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: > > > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 > > 192.168.147.130 > > s=FreeSWITCH > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/19a5dd14/attachment-0001.html From sz.krisz at freemail.hu Sun Jun 28 18:36:46 2009 From: sz.krisz at freemail.hu (szentesik) Date: Sun, 28 Jun 2009 18:36:46 -0700 (PDT) Subject: [Freeswitch-users] CTI In-Reply-To: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> Message-ID: <24247328.post@talk.nabble.com> Maxim Tsvetov wrote: > > Hello! > > We are seeking possibilities to use CTI features with Freeswitch. > > This features are: > - click-to-dial > - call popup > - answer call,hangup > - call transfer > > > Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, > CSTA..) > or there is already written module or third-party software? > ... > Currently working on some CSTA support (http://cstainside.sourceforge.net/). The MakeCall, DeliveredEvent, ClearConnection, TransferCall/SingleStepTransfer things required for the features above are on the list, the AnswerCall implementation is open (I'm not sure whether the FreeSWITCH is able to answer calls for any of the SIP clients available). -- View this message in context: http://www.nabble.com/CTI-tp24094686p24247328.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From fdenkens at ilibris.be Mon Jun 29 08:11:55 2009 From: fdenkens at ilibris.be (Frederik Denkens) Date: Mon, 29 Jun 2009 17:11:55 +0200 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation Message-ID: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> Hi, Following the recommendations of this list, we went with an external gateway to connect to the BRI based ISDN network. We'd like to configure a Patoon 4554 2xBRI <-> SIP gateway with Freeswitch to connect to the public ISDN network. The freeswitch config seems quite straightforward, but we don't manage to get the Patton to register with the Freeswitch and vise versa. Does anybody have some insight/sample config/tips they can share with us on this? Honesty obliges me to say that our experience with the Patton product is quite limited, which is not a big help for such a complex product. Many thanks in advance! Frederik. From msc at freeswitch.org Mon Jun 29 08:17:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 08:17:14 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24252099.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> Message-ID: <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> On Mon, Jun 29, 2009 at 4:01 AM, Edmar Cruz wrote: > > Yup. I got this mid_dahdi_codec but instructions how to install it? > Buy TC400B Install it and the TC400B drivers Load mod_dahdi_codec Enjoy low quality G729 calls. :) -MC > > Jason White-14 wrote: > > > > Edmar Cruz wrote: > >> > >> Is there any available license G729 for freeswitch? > > > > Yes. It was announced here a few days ago - see the list archives. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Is-there-any-license-G729--tp24251951p24252099.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/65d16b54/attachment.html From brian at freeswitch.org Mon Jun 29 08:19:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 10:19:16 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> Message-ID: Everyone has this need for lower bandwidth calls... I tend to march the other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a single ulaw call) /b On Jun 29, 2009, at 10:17 AM, Michael Collins wrote: > > > On Mon, Jun 29, 2009 at 4:01 AM, Edmar Cruz > wrote: > > Yup. I got this mid_dahdi_codec but instructions how to install it? > > Buy TC400B > Install it and the TC400B drivers > Load mod_dahdi_codec > Enjoy low quality G729 calls. :) > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/76e563d3/attachment.html From brian at freeswitch.org Mon Jun 29 08:20:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 10:20:22 -0500 Subject: [Freeswitch-users] CTI In-Reply-To: <24247328.post@talk.nabble.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> Message-ID: Nice are you the project leader? /b On Jun 28, 2009, at 8:36 PM, szentesik wrote: > Currently working on some CSTA support (http://cstainside.sourceforge.net/ > ). > The MakeCall, DeliveredEvent, ClearConnection, > TransferCall/SingleStepTransfer things required for the features > above are > on the list, the AnswerCall implementation is open (I'm not sure > whether the > FreeSWITCH is able to answer calls for any of the SIP clients > available). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/051eb4fd/attachment.html From mike at jerris.com Mon Jun 29 08:38:25 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:38:25 -0400 Subject: [Freeswitch-users] Result of an application... In-Reply-To: References: Message-ID: <548806F2-CE53-4D6F-89F8-C0EC7304C941@jerris.com> The application interface doesn't return a status like that. Different applications may set channel variables on an app by app basis. if your doing a bridge you can look at for example: http://wiki.freeswitch.org/wiki/Channel_Variables#originate_disposition Mike On Jun 27, 2009, at 7:47 PM, Klaus Hochlehnert wrote: > Hi, > > maybe a stupid question, but how can I find out the result of an > application? > > If I do (in perl) $session->execute("bridge", "user/${DestNr}\@ > $VARS{domain}"); > > How do I know if this was successful or if the user was busy or if > the phone doesn?t exist? > Is there any status variable for the result of an execute (or even > any other command)? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/93996e91/attachment.html From mike at jerris.com Mon Jun 29 08:40:24 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:40:24 -0400 Subject: [Freeswitch-users] hangup_cause NONE vs. NORMAL_CLEARING In-Reply-To: <1b46b4e80906280916g5fb065e9w65183f9117609f68@mail.gmail.com> References: <1b46b4e80906280916g5fb065e9w65183f9117609f68@mail.gmail.com> Message-ID: <68622E7A-8512-4631-8875-3745F85DC956@jerris.com> I would be curious if you have debug output of a call that is returning NONE there. That seems like we should be setting a hang-up cause somewhere. Mike On Jun 28, 2009, at 12:16 PM, Nicolas Brenner wrote: > I have a small JS script that makes a call, plays a sound file and > then hangs up. For each call it makes, I log the hangup_cause > variable on the CHANNEL_HANGUP_COMPLETE event. Most of the time, > when calls are successful, I get a NORMAL_CLEARING cause, but > sometimes I'll get a NONE cause. I wanted to know what the > difference between these two is, because there is no reference to > NONE in the wiki (http://wiki.freeswitch.org/wiki/Hangup_causes). > > Thanks, > > Nicol?s > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/2dd8a8be/attachment-0001.html From shiyanov at gmail.com Mon Jun 29 08:41:28 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 29 Jun 2009 19:41:28 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: Update again: FS debug logs of the problematic part http://pastebin.freeswitch.org/pastebin.php?dl=9512 Artem On Mon, Jun 29, 2009 at 7:07 PM, Artem Shiyanov wrote: > I've tried with the snapshot (06.26.2009) - and situation had become even > worse - now both agents hear nothing.. > Maybe problem is in my sip_profiles? > Here they are: > http://pastebin.freeswitch.org/pastebin.php?dl=9510 > http://pastebin.freeswitch.org/pastebin.php?dl=9511 > > > > > On Fri, Jun 26, 2009 at 10:03 PM, Brian West wrote: > >> Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk >> also... due to the lines below. >> >> /b >> >> On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: >> >> > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 >> > 192.168.147.130 >> > s=FreeSWITCH >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/0804e9af/attachment.html From mike at jerris.com Mon Jun 29 08:42:16 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:42:16 -0400 Subject: [Freeswitch-users] javascript FIFO (First In First Out) In-Reply-To: References: Message-ID: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> Fifo is not configured through javascript, it is configured via the configuration files. http://wiki.freeswitch.org/wiki/Mod_fifo Mike On Jun 29, 2009, at 12:53 AM, Baskar wrote: > Hi, > > I have configured inbound through JavaScript it is working well. > > Through dialplan i have configured FIFO it is also working fine but > i want to configure FIFO (First In First Out) through JavaScript. > > Is there any link or examples to configure FIFO through JavaScript > which will assist me to resolve this problem. > > can any one assist me to do this above process. > > Thanks in advance. > > Thanks with Regards, > N.Baskar. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/b4d33be5/attachment.html From william.suffill at gmail.com Mon Jun 29 08:42:52 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 29 Jun 2009 11:42:52 -0400 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> Message-ID: <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> You always have a way of beating to a different drum but whatever works. Good to see that the options are available and the user can choose what's best for their unique situation. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/3088057d/attachment.html From mike at jerris.com Mon Jun 29 08:43:54 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:43:54 -0400 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation In-Reply-To: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> References: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> Message-ID: Posting a quick sip trace of the failed register will probably be helpful. Also, are the debug logs any help? Mike On Jun 29, 2009, at 11:11 AM, Frederik Denkens wrote: > Hi, > > Following the recommendations of this list, we went with an external > gateway to connect to the BRI based ISDN network. > > We'd like to configure a Patoon 4554 2xBRI <-> SIP gateway with > Freeswitch to connect to the public ISDN network. The freeswitch > config seems quite straightforward, but we don't manage to get the > Patton to register with the Freeswitch and vise versa. > > Does anybody have some insight/sample config/tips they can share with > us on this? Honesty obliges me to say that our experience with the > Patton product is quite limited, which is not a big help for such a > complex product. From brian at freeswitch.org Mon Jun 29 08:44:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 10:44:52 -0500 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: <72397F6D-609C-43C4-998C-B5968506BF66@freeswitch.org> Now you'll need to outline step by step what you're doing to reproduce this problem. /b On Jun 29, 2009, at 10:41 AM, Artem Shiyanov wrote: > Update again: > FS debug logs of the problematic part > http://pastebin.freeswitch.org/pastebin.php?dl=9512 > > Artem > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/00784a0d/attachment.html From mnhassan at usa.net Mon Jun 29 08:48:34 2009 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 29 Jun 2009 22:48:34 +0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> Message-ID: <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> Hi, I just noted Micheal Collins mention "Enjoy lower quality G.729 calls". By "lower quality" do you mean G.729 in TC400B is lower in quality compared to software compression? Or is that comparing G.729 with G.711? Regards HASSAN On Mon, Jun 29, 2009 at 10:42 PM, William Suffill wrote: > You always have a way of beating to a different drum but whatever works. > > Good to see that the options are available and the user can choose what's > best for their unique situation. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From christian.loeschenkohl at xpirio.com Mon Jun 29 08:52:27 2009 From: christian.loeschenkohl at xpirio.com (=?UTF-8?B?Q2hyaXN0aWFuIEzDtnNjaGVua29obA==?=) Date: Mon, 29 Jun 2009 17:52:27 +0200 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation In-Reply-To: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> References: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> Message-ID: <4A48E33B.8040002@xpirio.com> hi i'm inalp patton certified, so maybe i can help if you can post your config (export startup config). please also describe your setup if i could get http+telnet access to the smartnode, the system would be registered in an minute br On 2009-06-29 17:11, Frederik Denkens wrote: > Hi, > > Following the recommendations of this list, we went with an external > gateway to connect to the BRI based ISDN network. > > We'd like to configure a Patoon 4554 2xBRI<-> SIP gateway with > Freeswitch to connect to the public ISDN network. The freeswitch > config seems quite straightforward, but we don't manage to get the > Patton to register with the Freeswitch and vise versa. > > Does anybody have some insight/sample config/tips they can share with > us on this? Honesty obliges me to say that our experience with the > Patton product is quite limited, which is not a big help for such a > complex product. > > Many thanks in advance! > > Frederik. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mike at jerris.com Mon Jun 29 09:00:25 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 12:00:25 -0400 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation In-Reply-To: <4A48E33B.8040002@xpirio.com> References: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> <4A48E33B.8040002@xpirio.com> Message-ID: If you come up with a good configuration, we would appretiate it if someone could post this to the wiki. Mike On Jun 29, 2009, at 11:52 AM, Christian L?schenkohl wrote: > hi > > i'm inalp patton certified, so maybe i can help > if you can post your config (export startup config). > please also describe your setup > if i could get http+telnet access to the smartnode, the system would > be registered in an minute > > br > From steveu at coppice.org Mon Jun 29 09:09:59 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 00:09:59 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> Message-ID: <4A48E757.60103@coppice.org> Nyamul Hassan wrote: > Hi, > > I just noted Micheal Collins mention "Enjoy lower quality G.729 > calls". By "lower quality" do you mean G.729 in TC400B is lower in > quality compared to software compression? Or is that comparing G.729 > with G.711? > I think he is just saddened by the way people tolerant crappy quality, and how slow the takeup of wideband voice has been. The TC400B doesn't do G.729. It does G.729A, which is significantly lower in quality. However, G.729A is what almost all equipment that vaguely says G.729 actually implements. Its lousy, but few people care. The TC400B works as well or as badly as anything else. Steve From jmesquita at gmail.com Mon Jun 29 09:22:04 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 29 Jun 2009 13:22:04 -0300 Subject: [Freeswitch-users] CTI In-Reply-To: References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> Message-ID: <5a8712120906290922q78515102t32e2f23dc4093015@mail.gmail.com> I am interested to know more about this. Are you using ESL to then translate CSTA calls to FS? Wouldn't this be great to be added as an FS module as an alternative to ESL? It would enable lots of existing CTI applications to work with FS. jmesquita On Mon, Jun 29, 2009 at 12:20 PM, Brian West wrote: > Nice are you the project leader? > /b > > On Jun 28, 2009, at 8:36 PM, szentesik wrote: > > Currently working on some CSTA support (http://cstainside.sourceforge.net/ > ). > The MakeCall, DeliveredEvent, ClearConnection, > TransferCall/SingleStepTransfer things required for the features above are > on the list, the AnswerCall implementation is open (I'm not sure whether > the > FreeSWITCH is able to answer calls for any of the SIP clients available). > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/03b422fe/attachment.html From mnhassan at usa.net Mon Jun 29 09:42:23 2009 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 29 Jun 2009 23:42:23 +0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48E757.60103@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> Message-ID: <9618647f0906290942s277ea8b2q6012b583d13d5f3e@mail.gmail.com> For wholesale termination setups, I think it makes it much more cost-effective to implement G.729 or even G.723. They can never be the same quality as G.711, but it saves bandwidth by multiple whole numbers, and is therefore, more sought after in markets where BW is expensive. One a side note, we were thinking of deploying a FS solution for our new retail platform. We are looking for a vendor who can provide a very good billing platform. Can you recommend me some? We are willing to pay for the service, but would rather go with an opensource project that provides paid support. Thank you in advance for your suggestions. Regards HASSAN On Mon, Jun 29, 2009 at 11:09 PM, Steve Underwood wrote: > Nyamul Hassan wrote: >> Hi, >> >> I just noted Micheal Collins mention "Enjoy lower quality G.729 >> calls". ?By "lower quality" do you mean G.729 in TC400B is lower in >> quality compared to software compression? ?Or is that comparing G.729 >> with G.711? >> > I think he is just saddened by the way people tolerant crappy quality, > and how slow the takeup of wideband voice has been. > > The TC400B doesn't do G.729. It does G.729A, which is significantly > lower in quality. However, G.729A is what almost all equipment that > vaguely says G.729 actually implements. Its lousy, but few people care. > The TC400B works as well or as badly as anything else. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From tayeb.meftah at gmail.com Mon Jun 29 09:46:53 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 29 Jun 2009 16:46:53 +0000 Subject: [Freeswitch-users] skypiax featur request Message-ID: <4A48EFFD.3080905@gmail.com> hello, this message is for skypiax developers / contributor please i have to request: i know a software named "SiSky" this is a SIP to Skype software gateway that let you interconnect your SIP IpPbx to the Skype network if i dial a skype speed dial number from my IpPhone, skype will no to bi visible but skypiax: if i dial any number i get skype visible and i heare the ringing sound in skype cool you try to hide it in the next release? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4196 (20090629) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From brian at freeswitch.org Mon Jun 29 09:49:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 11:49:27 -0500 Subject: [Freeswitch-users] skypiax featur request In-Reply-To: <4A48EFFD.3080905@gmail.com> References: <4A48EFFD.3080905@gmail.com> Message-ID: Please contact the author or open a jira as a feature request. /b On Jun 29, 2009, at 11:46 AM, Meftah Tayeb wrote: > hello, > this message is for skypiax developers / contributor > please i have to request: > i know a software named "SiSky" > this is a SIP to Skype software gateway that let you interconnect your > SIP IpPbx to the Skype network > if i dial a skype speed dial number from my IpPhone, skype will no > to bi > visible > but skypiax: if i dial any number i get skype visible and i heare the > ringing sound in skype > cool you try to hide it in the next release? > thanks From msc at freeswitch.org Mon Jun 29 09:52:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 09:52:33 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48E757.60103@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> Message-ID: <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> On Mon, Jun 29, 2009 at 9:09 AM, Steve Underwood wrote: > Nyamul Hassan wrote: > > Hi, > > > > I just noted Micheal Collins mention "Enjoy lower quality G.729 > > calls". By "lower quality" do you mean G.729 in TC400B is lower in > > quality compared to software compression? Or is that comparing G.729 > > with G.711? > > > I think he is just saddened by the way people tolerant crappy quality, > and how slow the takeup of wideband voice has been. > > The TC400B doesn't do G.729. It does G.729A, which is significantly > lower in quality. However, G.729A is what almost all equipment that > vaguely says G.729 actually implements. Its lousy, but few people care. > The TC400B works as well or as badly as anything else. > > Steve > Well said! Like Steve has pointed out in the past: G729/G729A is a race to the bottom. After using WB and UWB codecs all day every day for the past 6 months I just can't live with G729 or even GSM for that matter. However, to each his own. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/b1fc84e2/attachment.html From brian at freeswitch.org Mon Jun 29 10:00:09 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 12:00:09 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> Cellphones have really lowered everyones expectations of what quality is. I think if each of us in the US could stab our Cellphone providers in the neck and get away with it.. I know I would... YES AT&T I'm talking about your sorry excuse for service.... /b On Jun 29, 2009, at 11:52 AM, Michael Collins wrote: > Well said! Like Steve has pointed out in the past: G729/G729A is a > race to the bottom. After using WB and UWB codecs all day every day > for the past 6 months I just can't live with G729 or even GSM for > that matter. However, to each his own. > -MC > From steveu at coppice.org Mon Jun 29 10:11:51 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 01:11:51 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> Message-ID: <4A48F5D7.3050405@coppice.org> Brian West wrote: > Cellphones have really lowered everyones expectations of what quality > is. I think if each of us in the US could stab our Cellphone > providers in the neck and get away with it.. I know I would... YES > AT&T I'm talking about your sorry excuse for service.... > The 3G cellular standards have wideband features (that's what AMR-WB was developed for), but few (maybe no) operators enable them. I assume all UMTS phones actually support wideband, but unless the network is prepared to accept negotiation for it, phone support isn't much use. The networks would rather push proven flops like video calls. Steve From brian at freeswitch.org Mon Jun 29 10:19:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 12:19:16 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48F5D7.3050405@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> Message-ID: <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> Video calls would be great if any network would actually SUPPORT IT! (speaking US only) /b On Jun 29, 2009, at 12:11 PM, Steve Underwood wrote: > The 3G cellular standards have wideband features (that's what AMR-WB > was > developed for), but few (maybe no) operators enable them. I assume all > UMTS phones actually support wideband, but unless the network is > prepared to accept negotiation for it, phone support isn't much use. > The > networks would rather push proven flops like video calls. > > Steve From steveu at coppice.org Mon Jun 29 10:31:29 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 01:31:29 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> Message-ID: <4A48FA71.3050406@coppice.org> Video calls are a really really bad idea. People who think otherwise really haven't thought about it at all. They are available here, and people desperately don't want them to be. Brian West wrote: > Video calls would be great if any network would actually SUPPORT IT! > (speaking US only) > > /b > > On Jun 29, 2009, at 12:11 PM, Steve Underwood wrote: > > >> The 3G cellular standards have wideband features (that's what AMR-WB >> was >> developed for), but few (maybe no) operators enable them. I assume all >> UMTS phones actually support wideband, but unless the network is >> prepared to accept negotiation for it, phone support isn't much use. >> The >> networks would rather push proven flops like video calls. >> >> Steve Steve From brian at freeswitch.org Mon Jun 29 10:36:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 12:36:08 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48FA71.3050406@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> Message-ID: <50E06D00-0525-4BD2-8404-3654390EB0A9@freeswitch.org> Cuz its harder to lie to your wife! :P /b On Jun 29, 2009, at 12:31 PM, Steve Underwood wrote: > Video calls are a really really bad idea. People who think otherwise > really haven't thought about it at all. They are available here, and > people desperately don't want them to be. From msc at freeswitch.org Mon Jun 29 10:40:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 10:40:51 -0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Message-ID: <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> What kind of application are you building? Usually you want to use the dialplan to initiate the call and then let the js do the logical heavy lifting. -MC On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost wrote: > Dear All, > > I try > > s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); > if (s.ready()){ > s.setVariable("nibble_rate", "2.5"); > s.setVariable("nibble_account", "0838833133"); > s.execute("nibblebill", "heartbeat 5"); > bridge(session,s); > }; > > my question is > 1. How to cancel create s session (by dtmf ) like a * in bridge app > 2. when i hangup before s session ready is posible to cancel ? > > Best Regards. > > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a0e9273b/attachment.html From steveu at coppice.org Mon Jun 29 10:45:31 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 01:45:31 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <50E06D00-0525-4BD2-8404-3654390EB0A9@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <50E06D00-0525-4BD2-8404-3654390EB0A9@freeswitch.org> Message-ID: <4A48FDBB.80900@coppice.org> Brian West wrote: > Cuz its harder to lie to your wife! :P > > /b > > On Jun 29, 2009, at 12:31 PM, Steve Underwood wrote: > > >> Video calls are a really really bad idea. People who think otherwise >> really haven't thought about it at all. They are available here, and >> people desperately don't want them to be. >> Now you're thinking :-) Steve From Chr.Schaefers at gmx.de Mon Jun 29 10:51:43 2009 From: Chr.Schaefers at gmx.de (chschaef) Date: Mon, 29 Jun 2009 10:51:43 -0700 (PDT) Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? Message-ID: <24228390.post@talk.nabble.com> >Hi, >there is a "edit online" option to use to change the option on the phone. >You have to first install SIP Voip Settings Tool from the Nokia forum, and >then change the option "Secure call" to prefered from NCT. >Ognjen Hello, did you actually try it and succeed? The description in the Nokia forum is existing but the implementation of the listener is apparently missing ... regards, chschaef -- View this message in context: http://www.nabble.com/Anybody-tried-Nokia-E71-%28symbian-S60-3rd%29-with-Pjsip-and-TLS-SRTP--tp23230269p24228390.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 29 11:08:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 13:08:40 -0500 Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? In-Reply-To: <24228390.post@talk.nabble.com> References: <24228390.post@talk.nabble.com> Message-ID: You MUST use TLS btw for SRTP to work. I'm working right now to interop with my E63 /b On Jun 29, 2009, at 12:51 PM, chschaef wrote: > Hello, > > did you actually try it and succeed? The description in the Nokia > forum is > existing but the implementation of the listener is apparently > missing ... > > regards, > chschaef > -- From dave at 3c.co.uk Mon Jun 29 11:16:48 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 29 Jun 2009 21:16:48 +0300 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48FA71.3050406@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> Message-ID: <1246299408.3877.24.camel@dk-d820> On Tue, 2009-06-30 at 01:31 +0800, Steve Underwood wrote: > Video calls are a really really bad idea. People who think otherwise > really haven't thought about it at all. They are available here, and > people desperately don't want them to be. Video calls between 3G phones have been available in the UK for some years. I've only ever made two using one of these to real people, both across a table in a pub to show them what it looked like; I have made goodness knows how many to an IVR while trying to pick 3G-324M apart. However, there are some instances where they're very useful. For example, my family is geographically quite dispersed, and we use Skype with video a lot - particularly for grandparents to keep up with grandchildren. The usefulness and appropriateness of video calling depends very much on the target market; it's not, of itself, a bad thing. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From sz.krisz at freemail.hu Mon Jun 29 11:25:27 2009 From: sz.krisz at freemail.hu (szentesik) Date: Mon, 29 Jun 2009 11:25:27 -0700 (PDT) Subject: [Freeswitch-users] CTI In-Reply-To: <5a8712120906290922q78515102t32e2f23dc4093015@mail.gmail.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> <5a8712120906290922q78515102t32e2f23dc4093015@mail.gmail.com> Message-ID: <24259342.post@talk.nabble.com> The FS side CSTA "server" (mod_csta_socket) works as a loadable module exactly as you described, practically adding native CSTA support to it. (I test it only on Windows yet, but the code was made to let it build on any FS platform if someone put some effort into writing the makefiles etc.) Krisztian Jo?o Mesquita-3 wrote: > > I am interested to know more about this. Are you using ESL to then > translate > CSTA calls to FS? Wouldn't this be great to be added as an FS module as an > alternative to ESL? It would enable lots of existing CTI applications to > work with FS. > > jmesquita > > On Mon, Jun 29, 2009 at 12:20 PM, Brian West wrote: > >> Nice are you the project leader? >> /b >> >> On Jun 28, 2009, at 8:36 PM, szentesik wrote: >> >> Currently working on some CSTA support >> (http://cstainside.sourceforge.net/ >> ). > ... > -- View this message in context: http://www.nabble.com/CTI-tp24094686p24259342.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Jun 29 11:27:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 11:27:05 -0700 Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? In-Reply-To: References: <24228390.post@talk.nabble.com> Message-ID: <87f2f3b90906291127x60e630edu7c7d6b7992aca0c4@mail.gmail.com> On Mon, Jun 29, 2009 at 11:08 AM, Brian West wrote: > You MUST use TLS btw for SRTP to work. I'm working right now to > interop with my E63 > Sweet! Let us know when it's done. That's very cool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/164d5e40/attachment.html From Chr.Schaefers at gmx.de Mon Jun 29 11:36:16 2009 From: Chr.Schaefers at gmx.de (chschaef) Date: Mon, 29 Jun 2009 11:36:16 -0700 (PDT) Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? In-Reply-To: References: <49F2D874.7070002@gmx.net> <20090425095738.GA27179@jdc.jasonjgw.net> <49F341DF.20206@gmx.net> <4468a6770904270313l75bfd3fhc7406a526f001395@mail.gmail.com> <49F7138D.2060803@gmx.net> <4468a6770904280752s4f301ff6i5c966603b7b2d3f1@mail.gmail.com> <24228390.post@talk.nabble.com> Message-ID: <24259345.post@talk.nabble.com> Brian West-3 wrote: > >>You MUST use TLS btw for SRTP to work. I'm working right now to >> interop with my E63 >>/b > > I have configured TLS & SRTP on the Nokia E71 (and E51). That works fine to make encrypted calls. :-) When making a call to the Nokia, it does not respond on port 5061. On port 5060, it responds but only with UDP packets. Looking into the Nokia developer forum, I have found info from 2007, that the TLS listener is not yet implemented - but I could not get any confirmation by Nokia if that is still valid. So I appreciate that you get back to me if you can make the E63 calling securely in both directions. regards, chschaef -- View this message in context: http://www.nabble.com/Anybody-tried-Nokia-E71-%28symbian-S60-3rd%29-with-Pjsip-and-TLS-SRTP--tp23230269p24259345.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Mon Jun 29 12:47:23 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 20:47:23 +0100 Subject: [Freeswitch-users] bridge call from outbound socket Message-ID: Hi Guys, Is it possible to bridge to another destination while controlling a call via the outbound socket? In other words, I'm controlling a call using an outbound socket and at some point want to originate a new call leg and bridge the two. If it can't be done that way, I'm thinking I could originate the call using an inbound socket, grab the uuid and then call " api uuid_bridge " ? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/79a1b239/attachment-0001.html From msc at freeswitch.org Mon Jun 29 13:21:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 13:21:50 -0700 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: Message-ID: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> would you mind doing a pb of your script that is handling the OB event socket connection? -MC On Mon, Jun 29, 2009 at 12:47 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Is it possible to bridge to another destination while controlling a call > via the outbound socket? > > > > In other words, I?m controlling a call using an outbound socket and at some > point want to originate a new call leg and bridge the two. > > > > If it can?t be done that way, I?m thinking I could originate the call using an inbound socket, grab the uuid and then call ? api uuid_bridge ? ? > > > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/695273c7/attachment.html From sz.krisz at freemail.hu Mon Jun 29 13:34:42 2009 From: sz.krisz at freemail.hu (szentesik) Date: Mon, 29 Jun 2009 13:34:42 -0700 (PDT) Subject: [Freeswitch-users] CTI In-Reply-To: References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> Message-ID: <24261403.post@talk.nabble.com> Yes. It will also use to bring FS with a CTI application I'm a lead developer of closer, decided to make the integration open source/open standard based. Brian West-3 wrote: > > Nice are you the project leader? > > /b > > On Jun 28, 2009, at 8:36 PM, szentesik wrote: > >> Currently working on some CSTA support >> (http://cstainside.sourceforge.net/ >> ). >> The MakeCall, DeliveredEvent, ClearConnection, >> TransferCall/SingleStepTransfer things required for the features >> above are >> on the list, the AnswerCall implementation is open (I'm not sure >> whether the >> FreeSWITCH is able to answer calls for any of the SIP clients >> available). > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/CTI-tp24094686p24261403.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 29 13:38:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 15:38:20 -0500 Subject: [Freeswitch-users] CTI In-Reply-To: <24261403.post@talk.nabble.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> <24261403.post@talk.nabble.com> Message-ID: Are you interested in hosting any of it in our tree? /b On Jun 29, 2009, at 3:34 PM, szentesik wrote: > > Yes. It will also use to bring FS with a CTI application I'm a lead > developer > of closer, decided to make the integration open source/open standard > based. > From nik.middleton at noblesolutions.co.uk Mon Jun 29 13:43:54 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 21:43:54 +0100 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> Message-ID: Thanks, but I got it sorted. The bridge application along with the event-lock sorted. Works a treat, yippee! I'll write it up on the wiki Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 29 June 2009 21:22 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] bridge call from outbound socket would you mind doing a pb of your script that is handling the OB event socket connection? -MC On Mon, Jun 29, 2009 at 12:47 PM, Nik Middleton wrote: Hi Guys, Is it possible to bridge to another destination while controlling a call via the outbound socket? In other words, I'm controlling a call using an outbound socket and at some point want to originate a new call leg and bridge the two. If it can't be done that way, I'm thinking I could originate the call using an inbound socket, grab the uuid and then call " api uuid_bridge " ? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a17a2328/attachment.html From msc at freeswitch.org Mon Jun 29 13:49:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 13:49:23 -0700 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> Message-ID: <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Thanks, but I got it sorted. The bridge application along with the event-lock > sorted. Works a treat, yippee! I?ll write it up on the wiki > > > Cool deal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a6b733db/attachment.html From jmesquita at gmail.com Mon Jun 29 14:08:54 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 29 Jun 2009 18:08:54 -0300 Subject: [Freeswitch-users] CTI In-Reply-To: References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> <24261403.post@talk.nabble.com> Message-ID: <5a8712120906291408h54494e9fyde469901758bd757@mail.gmail.com> I would strongly suggest that. At least for the mod itself. That way, we can all contribute with it and keep it always compatible with the lib. jmesquita On Mon, Jun 29, 2009 at 5:38 PM, Brian West wrote: > Are you interested in hosting any of it in our tree? > > /b > > On Jun 29, 2009, at 3:34 PM, szentesik wrote: > > > > > Yes. It will also use to bring FS with a CTI application I'm a lead > > developer > > of closer, decided to make the integration open source/open standard > > based. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/eabe91ab/attachment.html From marketing at cluecon.com Mon Jun 29 14:33:43 2009 From: marketing at cluecon.com (Michael Collins) Date: Mon, 29 Jun 2009 14:33:43 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Last Call For Early Birds! Message-ID: <87f2f3b90906291433m4819f096nf38e20e7c0cc673f@mail.gmail.com> Good news! It's not too late to sign up for the early bird special of $499 for ClueCon, but you'll need to act fast - July 1 is the last day for the early bird special and it's only a few days away. If you haven't already registered for ClueCon then please call us and do so right away. Expedia.com still has deals on hotel rooms but the Wyndham is filling up quickly so don't delay in getting your room booked. In other news we have a new media sponsor: Biz-News.com. They will be conducting interviews with some of the speakers this year and are helping to promote the event. Please drop by their site and check them out! Looking forward to seeing everybody this August! -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/e3eb9e23/attachment-0001.html From nik.middleton at noblesolutions.co.uk Mon Jun 29 14:56:04 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 22:56:04 +0100 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> Message-ID: One little annoyance though, I cannot for the life of me get a ringback tone while the B leg is ringing, I've tried putting ringback=${us-ring} in the originate params, but no deal, just silence until the call is answered. Anyone care to shed some light on this? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 29 June 2009 21:49 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] bridge call from outbound socket On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton wrote: Thanks, but I got it sorted. The bridge application along with the event-lock sorted. Works a treat, yippee! I'll write it up on the wiki Cool deal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/ceb84138/attachment.html From msc at freeswitch.org Mon Jun 29 15:06:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 15:06:48 -0700 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> Message-ID: <87f2f3b90906291506n216db20du2de86d56593b96fe@mail.gmail.com> can you pb your script? -MC On Mon, Jun 29, 2009 at 2:56 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > One little annoyance though, I cannot for the life of me get a ringback > tone while the B leg is ringing, > > > > I?ve tried putting ringback=${us-ring} in the originate params, but no > deal, just silence until the call is answered. Anyone care to shed some > light on this? > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 29 June 2009 21:49 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] bridge call from outbound socket > > > > > > On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Thanks, but I got it sorted. The bridge application along with the event-lock > sorted. Works a treat, yippee! I?ll write it up on the wiki > > > > Cool deal. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/5be7eba8/attachment.html From mike at jerris.com Mon Jun 29 15:10:16 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 18:10:16 -0400 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> Message-ID: I don't think we expand vars here, you will need to expand us-ring yourself and pass the string Mike On Jun 29, 2009, at 5:56 PM, Nik Middleton wrote: > One little annoyance though, I cannot for the life of me get a > ringback tone while the B leg is ringing, > > I?ve tried putting ringback=${us-ring} in the originate params, but > no deal, just silence until the call is answered. Anyone care to > shed some light on this? > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: 29 June 2009 21:49 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] bridge call from outbound socket > > > On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton > wrote: > Thanks, but I got it sorted. The bridge application along with the > event-lock sorted. Works a treat, yippee! I?ll write it up on the > wiki > > > > Cool deal. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/9111f146/attachment.html From sz.krisz at freemail.hu Mon Jun 29 15:17:49 2009 From: sz.krisz at freemail.hu (=?ISO-8859-2?Q?Szentesi_Kriszti=E1n?=) Date: Tue, 30 Jun 2009 00:17:49 +0200 (CEST) Subject: [Freeswitch-users] CTI In-Reply-To: Message-ID: Yes. The project itself consist of 3 parts, the CSTAInsideCore static library for CSTA standard classes, the mod_csta_socket contains the FS specific stuff and a client program for learning purposes. Both the mod_csta_socket and the client links with the core library, all is standard C++. It is trivial to host the mod_csta_socket with FreeSWITCH, and would also agree to move the core library itself. Help from some experienced developers are always welcome. Krisztian Brian West ?rta: > Are you interested in hosting any of it in our tree? > > /b > > On Jun 29, 2009, at 3:34 PM, szentesik wrote: > > > > > Yes. It will also use to bring FS with a CTI application I'm a lead > > developer > > of closer, decided to make the integration open source/open standard > > based. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ________________________________________________________
Angol, n?met, spanyol, olasz, francia ny?ri intenz?v nyelvtanfolyamok indulnak j?nius 22-t?l a Bonus Nyelviskol?ban
http://ad.adverticum.net/b/cl,1,6022,335167,414321/click.prm -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/24ae026c/attachment-0001.html From msc at freeswitch.org Mon Jun 29 16:13:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 16:13:30 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 Message-ID: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> The FreeSWITCH development team is please to announce that there's a new version of FreeSWITCH available. Please update as soon as you reasonable can. More details available here: http://www.freeswitch.org/node/195 We appreciate everyone's help in making FreeSWITCH better. Please keep testing and reporting back! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/c761a15c/attachment.html From jason at jasonjgw.net Mon Jun 29 17:18:58 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 10:18:58 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> Message-ID: <20090630001858.GA12969@jdc.jasonjgw.net> Brian West wrote: > Everyone has this need for lower bandwidth calls... I tend to march the > other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a > single ulaw call) 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. This also works well in 48khz conferences. I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, G.722, G.722.1 and (my current favourite) Celt all the way. From dujinfang at gmail.com Mon Jun 29 18:13:20 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 09:13:20 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: > Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/ Linux/Windows, and I'd like to know some UAs supporting WB and UWB codecs. Thanks. > Well said! Like Steve has pointed out in the past: G729/G729A is a > race to the bottom. After using WB and UWB codecs all day every day > for the past 6 months I just can't live with G729 or even GSM for > that matter. However, to each his own. > -MC From sprice at gmail.com Mon Jun 29 18:18:16 2009 From: sprice at gmail.com (SP) Date: Mon, 29 Jun 2009 20:18:16 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> FreeSWITCH runs on a Mac and can be configured as a UA... also supports WB and UWB! On Mon, Jun 29, 2009 at 20:13, seven wrote: > > > > Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/ > Linux/Windows, and I'd like to know some UAs supporting WB and UWB > codecs. > > Thanks. > > > Well said! Like Steve has pointed out in the past: G729/G729A is a > > race to the bottom. After using WB and UWB codecs all day every day > > for the past 6 months I just can't live with G729 or even GSM for > > that matter. However, to each his own. > > -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/09c1d335/attachment.html From brian at freeswitch.org Mon Jun 29 18:21:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:21:25 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> Message-ID: oh and SUPER DUPER WIDE BAND. :P /b On Jun 29, 2009, at 8:18 PM, SP wrote: > FreeSWITCH runs on a Mac and can be configured as a UA... also > supports WB and UWB! From jason at jasonjgw.net Mon Jun 29 18:28:07 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 11:28:07 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> References: <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> Message-ID: <20090630012807.GA21806@jdc.jasonjgw.net> SP wrote: > FreeSWITCH runs on a Mac and can be configured as a UA... also supports WB > and UWB! Correct. It's especially good for those of us who prefer to avoid WIMP user interfaces. From craig at overthewire.com.au Mon Jun 29 18:26:15 2009 From: craig at overthewire.com.au (Craig Askings) Date: Tue, 30 Jun 2009 11:26:15 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Message-ID: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White : > Brian West wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! ?(btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From brian at freeswitch.org Mon Jun 29 18:31:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:31:39 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <20090630012807.GA21806@jdc.jasonjgw.net> References: <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> <20090630012807.GA21806@jdc.jasonjgw.net> Message-ID: <15355D4E-D2DE-4CAF-8DD1-8A587C7C6428@freeswitch.org> Its perfect for you isn't it... I rather love it too! ;) Simple... easy to use... and no fancy GUI to mess up... :P /b On Jun 29, 2009, at 8:28 PM, Jason White wrote: > Correct. It's especially good for those of us who prefer to avoid > WIMP user > interfaces. From dujinfang at gmail.com Mon Jun 29 18:36:43 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 09:36:43 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> Message-ID: <68966E85-CACE-4F69-8909-FE5744FB542C@gmail.com> Sure I know, Just haven't tried that. :) Any way it only makes sense when both legs support WB or UWB, unfortunately most of our b-legs are PSTN :(. > FreeSWITCH runs on a Mac and can be configured as a UA... also > supports WB and UWB! From darklion11 at yahoo.com Mon Jun 29 18:37:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 18:37:59 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <1246299408.3877.24.camel@dk-d820> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <1246299408.3877.24.camel@dk-d820> Message-ID: <24261922.post@talk.nabble.com> Please give me some free drivers of G729... Please David Knell wrote: > > On Tue, 2009-06-30 at 01:31 +0800, Steve Underwood wrote: >> Video calls are a really really bad idea. People who think otherwise >> really haven't thought about it at all. They are available here, and >> people desperately don't want them to be. > > Video calls between 3G phones have been available in the UK for some > years. I've only ever made two using one of these to real people, both > across a table in a pub to show them what it looked like; I have made > goodness knows how many to an IVR while trying to pick 3G-324M apart. > > However, there are some instances where they're very useful. For > example, my family is geographically quite dispersed, and we use Skype > with video a lot - particularly for grandparents to keep up with > grandchildren. The usefulness and appropriateness of video calling > depends very much on the target market; it's not, of itself, a bad > thing. > > --Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24261922.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Mon Jun 29 18:41:15 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 11:41:15 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> Message-ID: <20090630014115.GA23733@jdc.jasonjgw.net> Craig Askings wrote: > Are there any hardware phones that support 48 Khz Celt and > automated/mass deployment? Actually... FreeSWITCH in a phone could be a very good project. The main obstacles are: 1. Someone would need to design and build the hardware, or find existing hardware that would be suitable. 2. A script would have to be written to control i/o so that the keyboard and display of the phone could be used to make calls. From brian at freeswitch.org Mon Jun 29 18:43:28 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:43:28 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24261922.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <1246299408.3877.24.camel@dk-d820> <24261922.post@talk.nabble.com> Message-ID: <294FD821-2768-4545-BC9A-9DA875FAB27E@freeswitch.org> Free is not possible. /b PS: and its a codec :P On Jun 29, 2009, at 8:37 PM, Edmar Cruz wrote: > > Please give me some free drivers of G729... Please From brian at freeswitch.org Mon Jun 29 18:45:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:45:02 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <20090630014115.GA23733@jdc.jasonjgw.net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> Message-ID: <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> Now who can we con into building an open platform thats nothing more than a linux box shaped like a nice looking phone? :P Video and Touch Screen are a requirement :) /b On Jun 29, 2009, at 8:41 PM, Jason White wrote: > > Actually... FreeSWITCH in a phone could be a very good project. > > The main obstacles are: > > 1. Someone would need to design and build the hardware, or find > existing > hardware that would be suitable. > > 2. A script would have to be written to control i/o so that the > keyboard and > display of the phone could be used to make calls. From craig at overthewire.com.au Mon Jun 29 18:58:45 2009 From: craig at overthewire.com.au (Craig Askings) Date: Tue, 30 Jun 2009 11:58:45 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> Message-ID: <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> Well Snom phones use linux. I wonder if the the Chinese factory that pumps them out does grey market versions. Unfortunately that doesn't meed the video, touch screen or even the nice looking requirements. Craig. p.s. Above thoughts are my own not my employers. 2009/6/30 Brian West : > Now who can we con into building an open platform thats nothing more > than a linux box shaped like a nice looking phone? ?:P > > Video and Touch Screen are a requirement :) > > /b > > On Jun 29, 2009, at 8:41 PM, Jason White wrote: > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From sprice at gmail.com Mon Jun 29 19:23:03 2009 From: sprice at gmail.com (SP) Date: Mon, 29 Jun 2009 21:23:03 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> Message-ID: <7e2ac3270906291923ka1578fewa1ea07857147d1e6@mail.gmail.com> I'm now imagining bkw with his iphone in front of the mirror. - touchscreen - check - video - check - nice looking - you know I am On Mon, Jun 29, 2009 at 20:58, Craig Askings wrote: > Well Snom phones use linux. I wonder if the the Chinese factory that > pumps them out does grey market versions. Unfortunately that doesn't > meed the video, touch screen or even the nice looking requirements. > > Craig. > > p.s. Above thoughts are my own not my employers. > > 2009/6/30 Brian West : > > Now who can we con into building an open platform thats nothing more > > than a linux box shaped like a nice looking phone? :P > > > > Video and Touch Screen are a requirement :) > > > > /b > > > > On Jun 29, 2009, at 8:41 PM, Jason White wrote: > > > > -- > Craig Askings > > Network Engineer | Over the Wire Pty Ltd > craig at overthewire.com.au | www.overthewire.com.au > Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/e1cdf8e5/attachment.html From brian at freeswitch.org Mon Jun 29 19:30:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 21:30:11 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <7e2ac3270906291923ka1578fewa1ea07857147d1e6@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <7e2ac3270906291923ka1578fewa1ea07857147d1e6@mail.gmail.com> Message-ID: <0A6C17E1-6B63-44B4-8CBA-3339DD773030@freeswitch.org> iphone 3GS boi! /b On Jun 29, 2009, at 9:23 PM, SP wrote: > > I'm now imagining bkw with his iphone in front of the mirror. > touchscreen - check > video - check > nice looking - you know I am -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a218a045/attachment.html From drago at windstream.net Mon Jun 29 19:32:28 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 22:32:28 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> Message-ID: <019601c9f92b$036c02d0$0a440870$@net> Actually, Snom does have a version with color LCD touch screen - model 820. I'm not sure if it is in mass production yet. Regarding the "nice looking"... suit yourself :-) : http://www.snom.com/en/products/snom-820/ Drago -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Craig Askings Sent: Monday, June 29, 2009 9:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Well Snom phones use linux. I wonder if the the Chinese factory that pumps them out does grey market versions. Unfortunately that doesn't meed the video, touch screen or even the nice looking requirements. Craig. p.s. Above thoughts are my own not my employers. 2009/6/30 Brian West : > Now who can we con into building an open platform thats nothing more > than a linux box shaped like a nice looking phone? ?:P > > Video and Touch Screen are a requirement :) > > /b > > On Jun 29, 2009, at 8:41 PM, Jason White wrote: > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Jun 29 19:37:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 21:37:27 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <019601c9f92b$036c02d0$0a440870$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: BZZZT WRONG on the touch screen! /b On Jun 29, 2009, at 9:32 PM, Drago Totev wrote: > Actually, Snom does have a version with color LCD touch screen - > model 820. > I'm not sure if it is in mass production yet. > > Regarding the "nice looking"... suit yourself :-) : > http://www.snom.com/en/products/snom-820/ > > Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/d56708db/attachment.html From zhaoxxqq at 163.com Mon Jun 29 19:42:21 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Tue, 30 Jun 2009 10:42:21 +0800 Subject: [Freeswitch-users] Question about outband event use java Message-ID: <200906301042207132716@163.com> Hi folks, I'm really newbee to touch outband event. I write socket below and I write java server code like below: import java.io.BufferedReader; import java.io.BufferedWriter; import java.io.IOException; import java.io.InputStreamReader; import java.io.OutputStreamWriter; import java.io.PrintWriter; import java.net.ServerSocket; import java.net.Socket; import java.util.regex.*; public class ServerCode { /** * @param args */ public static int portNo = 8023; public static void main(String[] args) throws IOException { // TODO code application logic here ServerSocket s= new ServerSocket(portNo); System.out.println("The Server is start:" + s); Socket socket=s.accept(); try{ System.out.println("Accept the Client:"+ socket); BufferedReader in =new BufferedReader(new InputStreamReader(socket.getInputStream())); PrintWriter out = new PrintWriter(new BufferedWriter(new OutputStreamWriter(socket.getOutputStream())),true); while(true){ String str = in.readLine(); if(str.equals("byebye")){ break; } System.out.println("In Server recieved the info:" + str); out.println("connect\n\n"); } } finally { System.out.println("close the server socket and the io."); socket.close(); s.close(); } } } When I use extension 1001 to dial 3001, the cli display below: 2009-06-30 18:35:27 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 at 61.55.140.138:5060 [b9c80e50-6561-11de-8972-f3830035270b] 2009-06-30 18:35:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->3001 in context default I have send "connect\n\n" to FS, but no response to outband server. Can anyone help me? 2009-06-30 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/d7cc94f5/attachment-0001.html From drago at windstream.net Mon Jun 29 19:48:32 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 22:48:32 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: <01af01c9f92d$419ef1f0$c4dcd5d0$@net> Nope. J I "touch" it on VoiceCon in Orlando this April. Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 29, 2009 10:37 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? BZZZT WRONG on the touch screen! /b On Jun 29, 2009, at 9:32 PM, Drago Totev wrote: Actually, Snom does have a version with color LCD touch screen - model 820. I'm not sure if it is in mass production yet. Regarding the "nice looking"... suit yourself :-) : http://www.snom.com/en/products/snom-820/ Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/6ff46e85/attachment.html From jason at jasonjgw.net Mon Jun 29 19:50:28 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 12:50:28 +1000 Subject: [Freeswitch-users] Question about outband event use java In-Reply-To: <200906301042207132716@163.com> References: <200906301042207132716@163.com> Message-ID: <20090630025028.GA6448@jdc.jasonjgw.net> zhaoxxqq wrote: > and I write java server code like below: > > import java.io.BufferedReader; > import java.io.BufferedWriter; > import java.io.IOException; > import java.io.InputStreamReader; > import java.io.OutputStreamWriter; > import java.io.PrintWriter; > import java.net.ServerSocket; > import java.net.Socket; > import java.util.regex.*; Shouldn't you be using the Java event socket libraries that were discussed on the list recently? There's also one in the FreeSWITCH source tree now. From brian at freeswitch.org Mon Jun 29 19:51:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 21:51:18 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <01af01c9f92d$419ef1f0$c4dcd5d0$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <01af01c9f92d$419ef1f0$c4dcd5d0$@net> Message-ID: <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> Yah you can touch the screen all you want on the 820 but mine doesn't do jack! /b On Jun 29, 2009, at 9:48 PM, Drago Totev wrote: > Nope. J > > I ?touch? it on VoiceCon in Orlando this April. > > Drago > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/54b9fc35/attachment.html From hads at nice.net.nz Mon Jun 29 19:55:32 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 30 Jun 2009 14:55:32 +1200 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: <1246330532.4148.3.camel@sodium> On Mon, 2009-06-29 at 21:37 -0500, Brian West wrote: > BZZZT WRONG on the touch screen! Just wrong on the model, it's the 870 which has a touch screen. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From drago at windstream.net Mon Jun 29 19:56:07 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 22:56:07 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <01af01c9f92d$419ef1f0$c4dcd5d0$@net> <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> Message-ID: <01c001c9f92e$5150ee40$f3f2cac0$@net> Correct. As I said - it is a prototype. We wasting time in this argument. Contact Snom if you are interested or. move on, buddy J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 29, 2009 10:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Yah you can touch the screen all you want on the 820 but mine doesn't do jack! /b On Jun 29, 2009, at 9:48 PM, Drago Totev wrote: Nope. J I "touch" it on VoiceCon in Orlando this April. Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/101d5857/attachment-0001.html From drago at windstream.net Mon Jun 29 20:01:05 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 23:01:05 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <1246330532.4148.3.camel@sodium> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> Message-ID: <01cb01c9f92f$029024f0$07b06ed0$@net> You are correct. My bad. Drago -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hadley Rich Sent: Monday, June 29, 2009 10:56 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? On Mon, 2009-06-29 at 21:37 -0500, Brian West wrote: > BZZZT WRONG on the touch screen! Just wrong on the model, it's the 870 which has a touch screen. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Jun 29 20:03:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 22:03:21 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <1246330532.4148.3.camel@sodium> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> Message-ID: <3FCFEAE4-94FC-45FD-813D-535896DE5ADD@freeswitch.org> I'll have to call Christian over at Snom and have him ship me one then! ;) /b On Jun 29, 2009, at 9:55 PM, Hadley Rich wrote: > Just wrong on the model, it's the 870 which has a touch screen. > > hads From brian at freeswitch.org Mon Jun 29 20:03:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 22:03:40 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <01c001c9f92e$5150ee40$f3f2cac0$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <01af01c9f92d$419ef1f0$c4dcd5d0$@net> <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> <01c001c9f92e$5150ee40$f3f2cac0$@net> Message-ID: <7AD463C5-0A80-4625-81A1-038ED9EBB47E@freeswitch.org> I'll have to get one now... I have one of every other snom! :) /b On Jun 29, 2009, at 9:56 PM, Drago Totev wrote: > Correct. As I said ? it is a prototype. > > We wasting time in this argument. Contact Snom if you are interested > or? move on, buddy J > > Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/f70077fe/attachment.html From brian at freeswitch.org Mon Jun 29 20:04:12 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 22:04:12 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <01cb01c9f92f$029024f0$07b06ed0$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> <01cb01c9f92f$029024f0$07b06ed0$@net> Message-ID: Its ok ... I have lysdexic moments almost daily. /b On Jun 29, 2009, at 10:01 PM, Drago Totev wrote: > You are correct. My bad. > > Drago From drago at windstream.net Mon Jun 29 20:07:44 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 23:07:44 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> <01cb01c9f92f$029024f0$07b06ed0$@net> Message-ID: <01d201c9f92f$f054d3c0$d0fe7b40$@net> Now I feel better! I thought it's me only... :-) Drago -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 29, 2009 11:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Its ok ... I have lysdexic moments almost daily. /b On Jun 29, 2009, at 10:01 PM, Drago Totev wrote: > You are correct. My bad. > > Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Mon Jun 29 20:08:40 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 11:08:40 +0800 Subject: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h Message-ID: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> Hi, I'm on the latest svn 14041, and I have tiff installed with "port install tiff", how can I tell FS to find libtiff at /opt/local/include? on Linux the package should be libtiff-dev of libtiff-devel, but I think tiff is the equivalent on Mac. previous versions build ok on my Mac, possible to include the libtiff in trunk like other libs? checking tiffio.h usability... no checking tiffio.h presence... no checking for tiffio.h... no checking pthread.h usability... yes checking pthread.h presence... yes checking for pthread.h... yes checking X11/X.h usability... yes checking X11/X.h presence... yes checking for X11/X.h... yes checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h usability... yes checking libxml/xmlmemory.h presence... yes checking for libxml/xmlmemory.h... yes checking libxml/parser.h usability... yes checking libxml/parser.h presence... yes checking for libxml/parser.h... yes checking libxml/xinclude.h usability... yes checking libxml/xinclude.h presence... yes checking for libxml/xinclude.h... yes checking FL/Fl.H usability... no checking FL/Fl.H presence... no checking for FL/Fl.H... no checking FL/Fl_Overlay_Window.H usability... no checking FL/Fl_Overlay_Window.H presence... no checking for FL/Fl_Overlay_Window.H... no checking FL/Fl_Light_Button.H usability... no checking FL/Fl_Light_Button.H presence... no checking for FL/Fl_Light_Button.H... no checking FL/fl_draw.H usability... no checking FL/fl_draw.H presence... no checking for FL/fl_draw.H... no checking FL/Fl_Cartesian.H usability... no checking FL/Fl_Cartesian.H presence... no checking for FL/Fl_Cartesian.H... no checking FL/Fl_Audio_Meter.H usability... no checking FL/Fl_Audio_Meter.H presence... no checking for FL/Fl_Audio_Meter.H... no checking for cos in -lm... yes checking for library containing sinf... none required checking for library containing cosf... none required checking for library containing tanf... none required checking for library containing asinf... none required checking for library containing acosf... none required checking for library containing atanf... none required checking for library containing atan2f... none required checking for library containing ceilf... none required checking for library containing floorf... none required checking for library containing powf... none required checking for library containing expf... none required checking for library containing logf... none required checking for library containing log10f... none required checking for TIFFOpen in -ltiff... no configure: error: "Can't build without libtiff (does your system require a libtiff-devel package?)" configure: error: ./configure.gnu failed for libs/spandsp From craig at overthewire.com.au Mon Jun 29 20:20:18 2009 From: craig at overthewire.com.au (Craig Askings) Date: Tue, 30 Jun 2009 13:20:18 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <1246330532.4148.3.camel@sodium> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> Message-ID: <8cc991dd0906292020u3528f483v1462e049877c664d@mail.gmail.com> Well the 8XX series is much prettier than the 3XX series. Though I wonder if anyone has thought of porting FreeSWITCH to the Palm Pre? Craig. 2009/6/30 Hadley Rich : > On Mon, 2009-06-29 at 21:37 -0500, Brian West wrote: >> BZZZT WRONG on the touch screen! > > Just wrong on the model, it's the 870 which has a touch screen. > > hads > -- > http://nicegear.co.nz > New Zealand's Open Source Hardware Supplier > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From diego.viola at gmail.com Mon Jun 29 20:26:36 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 29 Jun 2009 23:26:36 -0400 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <20090629105113.GA4756@jdc.jasonjgw.net> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> Message-ID: <86a32abc0906292026v27065787ida78f6b2ca0afb3c@mail.gmail.com> Look for this email: "[Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH" I think the development team is also about to offer G729 licensing if I'm not mistaken. On Mon, Jun 29, 2009 at 6:51 AM, Jason White wrote: > Edmar Cruz wrote: > > > > Is there any available license G729 for freeswitch? > > Yes. It was announced here a few days ago - see the list archives. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/34d1d2e5/attachment.html From vince.freeswitch at hightek.org Mon Jun 29 21:19:12 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Mon, 29 Jun 2009 23:19:12 -0500 Subject: [Freeswitch-users] freeswitch on Dragonfly BSD (RLIMIT_AS issue) In-Reply-To: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> Message-ID: <20090630041912.GA2585@quark.hightek.org> On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > > Can we make a patch ifdefing on RLIMIT_AS to make this always work > without patches to system header files? I have attached a patch to src/switch_core.c that fixes this. -------------- next part -------------- --- src/switch_core.c.orig 2009-06-29 19:29:22 -0700 +++ src/switch_core.c 2009-06-29 19:31:21 -0700 @@ -826,7 +826,9 @@ setrlimit(RLIMIT_RTPRIO, &rlp); #endif -#if !defined(__OpenBSD__) && !defined(__NetBSD__) +#if defined(__DragonFly__) + setrlimit(RLIMIT_VMEM, &rlp); +#elif !defined(__OpenBSD__) && !defined(__NetBSD__) setrlimit(RLIMIT_AS, &rlp); #endif #endif From vince.freeswitch at hightek.org Mon Jun 29 21:35:34 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Mon, 29 Jun 2009 23:35:34 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090623040853.GA84157@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> Message-ID: <20090630043534.GA2840@quark.hightek.org> Update: I tested compiling on my Dragonfly leaf development account which is running DragonFly 2.2.1-RELEASE with gcc-4.1.2. I did not get any of the cast warnings. It is apparently a gcc issue since I am running gcc-3.4.6 on my workstation with an older Dragonfly installation. Apparently gcc-3 does not like the "return x, y;" syntax that is used all over the sofia code. That is the source of most of the warnings (except 1 or 2 so far that just needed a cast). So, on DF 2.2.1, it appeared to compile almost successfully until near the end. It still has that sym link error and still has the build bug where it does not know there was an error and goes on to tell you it was successful. Here is the final output: making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /home/vince/freeswitch/freeswitch-20090623/libs/js/nsprpub/dist/include/nspr/.: Invalid argument gmake[9]: *** [export] Error 1 gmake[8]: *** [export] Error 2 gmake[7]: *** [export] Error 2 gmake[6]: *** [export] Error 2 gmake[5]: *** [/home/vince/freeswitch/freeswitch-20090623/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 From yj13535428332 at gmail.com Mon Jun 29 21:03:48 2009 From: yj13535428332 at gmail.com (jun yang) Date: Tue, 30 Jun 2009 12:03:48 +0800 Subject: [Freeswitch-users] how to record the conference manually? Message-ID: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> Hi,everyone. new to play with freeswitch. i have setup freeswitch and can call each other very well. now, a conference 3002 with several users in it. i want to record 3002 manually, but can't get the way. i have try fs_cli use the command: conference 3002 record /tmp/foo.wav it response: conference 3002 not found any clue? thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/faf33374/attachment.html From brian at freeswitch.org Mon Jun 29 22:43:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 30 Jun 2009 00:43:30 -0500 Subject: [Freeswitch-users] how to record the conference manually? In-Reply-To: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> References: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> Message-ID: I suspect the conference is number at domain... do a conference list and check. /b On Jun 29, 2009, at 11:03 PM, jun yang wrote: > Hi,everyone. new to play with freeswitch. > i have setup freeswitch and can call each other very well. > now, a conference 3002 with several users in it. > i want to record 3002 manually, but can't get the way. > i have try fs_cli use the command: > conference 3002 record /tmp/foo.wav > it response: conference 3002 not found > any clue? > thanks in advance! From jason at jasonjgw.net Mon Jun 29 22:50:08 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 15:50:08 +1000 Subject: [Freeswitch-users] how to record the conference manually? In-Reply-To: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> References: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> Message-ID: <20090630055008.GA2693@jdc.jasonjgw.net> jun yang wrote: > now, a conference 3002 with several users in it. > i want to record 3002 manually, but can't get the way. > i have try fs_cli use the command: > conference 3002 record /tmp/foo.wav > it response: conference 3002 not found You need to specify the full conference name, including the domain. conference list will show you the full names of active conferences. From dome at tel.co.th Mon Jun 29 23:07:15 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 30 Jun 2009 13:07:15 +0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> Message-ID: <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> 2009/6/30 Michael Collins : > What kind of application are you building? Usually you want to use the > dialplan to initiate the call and then let the js do the logical heavy > lifting. I'm use js for callback solution. Dome C. > > -MC > > On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost wrote: >> >> Dear All, >> >> I try >> >> s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); >> if (s.ready()){ >> ? s.setVariable("nibble_rate", "2.5"); >> ? s.setVariable("nibble_account", "0838833133"); >> ? s.execute("nibblebill", "heartbeat 5"); >> ? bridge(session,s); >> }; >> >> my question is >> 1. How to cancel create s session (by dtmf ) like a * in bridge app >> 2. when i hangup before s session ready is posible to cancel ? >> >> Best Regards. >> >> Dome C. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Jun 29 23:17:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 23:17:42 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24261922.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <1246299408.3877.24.camel@dk-d820> <24261922.post@talk.nabble.com> Message-ID: <87f2f3b90906292317l31e7f64dkc22fd0e9b2b0bd8a@mail.gmail.com> On Mon, Jun 29, 2009 at 6:37 PM, Edmar Cruz wrote: > > Please give me some free drivers of G729... Please > This is the primary reason that most OSS guys hate G729 - it's patent encumbered BIG TIME. Check out this article on the main FS site: http://www.freeswitch.org/node/155 look at the patent claims on G729. G729A is even worse. Add to that the fact that the voice quality is poor and you have a recipe for a codec that telecom geeks don't really care for all that much. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/df16aa4e/attachment.html From msc at freeswitch.org Mon Jun 29 23:19:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 23:19:31 -0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> Message-ID: <87f2f3b90906292319v3d5dcee7ub2820097ff52f551@mail.gmail.com> can you post your script and dialplan? Let's take a look. -MC On Mon, Jun 29, 2009 at 11:07 PM, Dome Charoenyost wrote: > 2009/6/30 Michael Collins : > > What kind of application are you building? Usually you want to use the > > dialplan to initiate the call and then let the js do the logical heavy > > lifting. > I'm use js for callback solution. > > Dome C. > > > > > -MC > > > > On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost > wrote: > >> > >> Dear All, > >> > >> I try > >> > >> s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); > >> if (s.ready()){ > >> s.setVariable("nibble_rate", "2.5"); > >> s.setVariable("nibble_account", "0838833133"); > >> s.execute("nibblebill", "heartbeat 5"); > >> bridge(session,s); > >> }; > >> > >> my question is > >> 1. How to cancel create s session (by dtmf ) like a * in bridge app > >> 2. when i hangup before s session ready is posible to cancel ? > >> > >> Best Regards. > >> > >> Dome C. > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/0e3f2e46/attachment.html From yj13535428332 at gmail.com Mon Jun 29 23:22:04 2009 From: yj13535428332 at gmail.com (jun yang) Date: Tue, 30 Jun 2009 14:22:04 +0800 Subject: [Freeswitch-users] how to record the conference manually? In-Reply-To: <20090630055008.GA2693@jdc.jasonjgw.net> References: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> <20090630055008.GA2693@jdc.jasonjgw.net> Message-ID: <6536698d0906292322t54a63549q266f05e10b89c1be@mail.gmail.com> thanks,Brian West and Jason White. now it works fine. On Tue, Jun 30, 2009 at 1:50 PM, Jason White wrote: > jun yang wrote: > > now, a conference 3002 with several users in it. > > i want to record 3002 manually, but can't get the way. > > i have try fs_cli use the command: > > conference 3002 record /tmp/foo.wav > > it response: conference 3002 not found > > You need to specify the full conference name, including the domain. > > conference list > will show you the full names of active conferences. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/33eaf6f9/attachment-0001.html From darklion11 at yahoo.com Mon Jun 29 23:26:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 23:26:04 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: <24266762.post@talk.nabble.com> I need actually G729 license if there's any for freeswitch to call mobile phones... I already load it but has an issue on it passthrough mode... If i install mod_dandhi_codec it overwrites the existing G729 without license to a new G729 with license? dujinfang wrote: > >> > > Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/ > Linux/Windows, and I'd like to know some UAs supporting WB and UWB > codecs. > > Thanks. > >> Well said! Like Steve has pointed out in the past: G729/G729A is a >> race to the bottom. After using WB and UWB codecs all day every day >> for the past 6 months I just can't live with G729 or even GSM for >> that matter. However, to each his own. >> -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24266762.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brad.tuan at gmail.com Mon Jun 29 23:26:09 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 30 Jun 2009 14:26:09 +0800 Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? Message-ID: I have already set it in my FreeSwitch\conf\sip_profiles\external.xml: But my FS_A always return a 403 to FS_B, Where is the problem?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/433e7289/attachment.html From darklion11 at yahoo.com Mon Jun 29 23:37:27 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 23:37:27 -0700 (PDT) Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? In-Reply-To: References: Message-ID: <24266862.post@talk.nabble.com> Create a dialplan for FS A to communicate to FS B. dialplan/default/00_fsa.xml -> IP of FS B To check if your gateway succeed just type on sofia status check if gateway FreeSWITCH is there connection is set and ready to go :) Brad Tuan wrote: > > I have already set it in my FreeSwitch\conf\sip_profiles\external.xml: > > > > > > > > > > > > > But my FS_A always return a 403 to FS_B, Where is the problem?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-set-FS_A-as-a-gateway-of-FS_B---tp24266777p24266862.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yudha2008 at gmail.com Mon Jun 29 23:54:36 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 30 Jun 2009 12:24:36 +0530 Subject: [Freeswitch-users] javascript FIFO (First In First Out) In-Reply-To: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> References: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> Message-ID: *Hi Michael Jerris, Is there any other possible way to queue the inbound call in JavaScript. I am working on this process: step 1: I want the inbound call to be in queue through JavaScript step2: Then JavaScript will check most waiting agent and bridge the call to the most waiting agent ( this has been done in JavaScript itself) I want the incoming call to be in queue Is this possible through java script. Assist me to resolve this problem. Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/330a5a5b/attachment.html From dome at tel.co.th Tue Jun 30 00:01:23 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 30 Jun 2009 14:01:23 +0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <87f2f3b90906292319v3d5dcee7ub2820097ff52f551@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> <87f2f3b90906292319v3d5dcee7ub2820097ff52f551@mail.gmail.com> Message-ID: <8ccbff060906300001wef058c6m5edf3eafab9a57c0@mail.gmail.com> 2009/6/30 Michael Collins : > can you post your script and dialplan? Let's take a look. > -MC dialstr[i] is array like a sofia/external/1111 at xxx.xxx.xxx.xxx dial_option = "{absolute_codec_string='GSM,G729',ignore_early_media=false,originate_timeout=20,origination_caller_id=xxxxx} for (var i = 1; i <= 3; i++){ if (session.ready()){ session.execute("set", "hangup_after_bridge=false"); result = session.setAutoHangup(false); s1 = new Session(dial_option+dialstr[i]); } if (s1.ready()){ s1.setVariable("nibble_rate", "2.5"); s1.execute("set", "hangup_after_bridge=false"); s1.setVariable("nibble_account", acaller); s1.setVariable("provider_id", dialprovider_id[i]); s1.setVariable("provider", dialprovider[i]); s1.setVariable("service_charge", dialservice_charge[i]); s1.execute("set", "destination_number="+number); s1.execute("nibblebill", "heartbeat 5"); bridge(session,s1); console_log("notice", "Disconnect cause: " + s1.cause + " Code:"+s1.causecode+"\n"); }; if (s1.causecode==16 || s1.causecode==0){ i =10; }; }; > > On Mon, Jun 29, 2009 at 11:07 PM, Dome Charoenyost wrote: >> >> 2009/6/30 Michael Collins : >> > What kind of application are you building? Usually you want to use the >> > dialplan to initiate the call and then let the js do the logical heavy >> > lifting. >> I'm use js for callback solution. >> >> Dome C. >> >> > >> > -MC >> > >> > On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost >> > wrote: >> >> >> >> Dear All, >> >> >> >> I try >> >> >> >> s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); >> >> if (s.ready()){ >> >> ? s.setVariable("nibble_rate", "2.5"); >> >> ? s.setVariable("nibble_account", "0838833133"); >> >> ? s.execute("nibblebill", "heartbeat 5"); >> >> ? bridge(session,s); >> >> }; >> >> >> >> my question is >> >> 1. How to cancel create s session (by dtmf ) like a * in bridge app >> >> 2. when i hangup before s session ready is posible to cancel ? >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brad.tuan at gmail.com Tue Jun 30 00:05:28 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 30 Jun 2009 15:05:28 +0800 Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? Message-ID: I know that i need to set the dialplan, my problem is when FS_B send a REGISTER to FS_A, FS_A will return a 403 to FS_B Like this: 2009-06-30 15:03:25 [NOTICE] sofia_reg.c:305 sofia_reg_check_gateway() Registering FS_A 2009-06-30 15:03:25 [ERR] sofia_reg.c:1391 sofia_reg_handle_sip_r_register() FS_A Registration Failed with status Forbidden [403]. failure #2 2009-06-30 15:03:25 [WARNING] sofia_reg.c:334 sofia_reg_check_gateway() FS_A Failed Registration, setting retry to 30 seconds. > Create a dialplan for FS A to communicate to FS B. > > dialplan/default/00_fsa.xml > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > data="sofia/gateway/FreeSwitch/1$218.210.xxx.xxx"/> -> IP of FS B > > > To check if your gateway succeed > just type on sofia status check if gateway FreeSWITCH is there connection is > set and ready to go :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/38ca157f/attachment.html From dujinfang at gmail.com Tue Jun 30 01:25:35 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 16:25:35 +0800 Subject: [Freeswitch-users] javascript FIFO (First In First Out) In-Reply-To: References: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> Message-ID: <4BC016E6-75F0-48C6-B305-6D4C19799CF3@gmail.com> On Jun 30, 2009, at 2:54 PM, Baskar wrote: > Hi Michael Jerris, > > Is there any other possible way to queue the inbound call in > JavaScript. > > I am working on this process: > > step 1: I want the inbound call to be in queue through JavaScript > > step2: Then JavaScript will check most waiting agent and bridge the > call to the most waiting agent ( this has been done in JavaScript > itself) Just curious, why use a js while mod_fifo can do ? > > > I want the incoming call to be in queue Is this possible through > java script. > > Assist me to resolve this problem. > > > > Thanks with Regards, > N.Baskar > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/f03967c8/attachment.html From saeedahmad1981 at gmail.com Tue Jun 30 02:04:30 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 30 Jun 2009 11:04:30 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: Hi, What is the best way to update to latest version if we are already running an older stable version? I am using SVN trunk, sources are in /usr/src and its installed in /usr/local/freeswitch >From the same src directory, is it possible to install latest rev in somewhere else (for example /opt/freeswitch), if everything is working ok then i replace it in /usr/local/freeswitch? or any other alternatives? Many thanks. On Tue, Jun 30, 2009 at 1:13 AM, Michael Collins wrote: > The FreeSWITCH development team is please to announce that there's a new > version of FreeSWITCH available. Please update as soon as you reasonable > can. More details available here: > http://www.freeswitch.org/node/195 > > We appreciate everyone's help in making FreeSWITCH better. Please keep > testing and reporting back! > > -Michael > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/9753e692/attachment-0001.html From fdelawarde at wirelessmundi.com Tue Jun 30 02:15:49 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 30 Jun 2009 11:15:49 +0200 Subject: [Freeswitch-users] Any advances on T.38 support for FS? Message-ID: <1246353349.30167.83.camel@luna.tc.commsmundi.com> Many issues on Asterisk's T.38 (or probably just on T.38?)... Could it convince those relying on this "modern" version of a 50yo technology to switch to and with FreeSwitch? Fran?ois. From jason at jasonjgw.net Tue Jun 30 02:14:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 19:14:56 +1000 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: <20090630091456.GA5371@jdc.jasonjgw.net> Saeed Ahmad wrote: > What is the best way to update to latest version if we are already running > an older stable version? You did ask for the best way, which is to build packages for your operating system, then use your operating system's package manager to install them and keep track of different versions. This way, you can be sure that the right files are installed, that old versions are cleanly deleted (unless there's an error in the post-installation script, in which case it's a bug) and you can use the package manager to find out what files are installed and where they reside. You can then upgrade or downgrade simply by installing a different version of the package. FreeSWITCH supports building Debian packages, and there is also support for Centos and Fedora. From yudha2008 at gmail.com Tue Jun 30 02:27:20 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 30 Jun 2009 14:57:20 +0530 Subject: [Freeswitch-users] Error in OpenZap Message-ID: *Hi, **I have installed the latest trunk with A102D Sangoma card when i load the openzap i get this error ** freeswitch at localhost.localdomain> load mod_openzap 2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules configured: 1 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '1' API CALL [load(mod_openzap)] output: -ERR [module load file routine returned an error] 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:1-15' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '1:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:17-31' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '2' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:1-15' freeswitch at localhost.localdomain> 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:17-31' 2009-06-30 14:47:33 [INFO] zap_io.c:2370 load_config() Configured 0 channel(s) 2009-06-30 14:47:33 [ERR] zap_io.c:2633 zap_global_init() No modules configured! 2009-06-30 14:47:33 [ERR] mod_openzap.c:2401 mod_openzap_load() Error loading OpenZAP 2009-06-30 14:47:33 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so **Module load routine returned an error** How can i resolve the above problem. how can i load the openzap My Configuration files default.xml /etc/wanpipe/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 0 PCIBUS = 2 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml /usr/local/freeswitch/libs/openzap/conf/openzap.conf [span wanpipe1] number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 **How can i resolve the above problem. how can i load the openzap. can some one assist me to resolve this process.* * -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/ca34bb7e/attachment.html From raul at etellicom.com Tue Jun 30 02:37:41 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 30 Jun 2009 06:37:41 -0300 Subject: [Freeswitch-users] Error in OpenZap In-Reply-To: References: Message-ID: <1246354661.19445.13.camel@raul-laptop> Replace [span wanpipe1] with [span wanpipe 1] in your openzap.conf file - it's missing a space between wanpipe (the span type) and 1 (the span ID). Regards, Raul On Tue, 2009-06-30 at 14:57 +0530, Baskar wrote: > Hi, > > I have installed the latest trunk with A102D Sangoma card when i load > the openzap i get this error > > freeswitch at localhost.localdomain> load mod_openzap > 2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules > configured: 1 > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'name' / 'OpenZAP' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'number' / '1' > API CALL [load(mod_openzap)] output: > -ERR [module load file routine returned an error] > > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'trunk_type' / 'E1' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '1:1-15' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'd-channel' / '1:16' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '1:17-31' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'name' / 'OpenZAP' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'number' / '2' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'trunk_type' / 'E1' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '2:1-15' > freeswitch at localhost.localdomain> 2009-06-30 14:47:33 [ERR] > zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '2:17-31' > 2009-06-30 14:47:33 [INFO] zap_io.c:2370 load_config() Configured 0 > channel(s) > 2009-06-30 14:47:33 [ERR] zap_io.c:2633 zap_global_init() No modules > configured! > 2009-06-30 14:47:33 [ERR] mod_openzap.c:2401 mod_openzap_load() Error > loading OpenZAP > 2009-06-30 14:47:33 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading > module /usr/local/freeswitch/mod/mod_openzap.so > **Module load routine returned an error** > > How can i resolve the above problem. how can i load the openzap > > My Configuration files > > > default.xml > > > expression="^9(\d{5,15})$"> > > data="OpenZAP/1/a/${dialed_ext}"/> > > > > /etc/wanpipe/wanpipe1.conf > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 0 > PCIBUS = 2 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = CRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 120 > LBO = 120OH > TE_SIG_MODE = CCS > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 16 > TDMV_HW_DTMF = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_ECHO_OFF = NO > TDMV_HWEC = NO > > > /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml > > > > > > > > > > > > > > > > /usr/local/freeswitch/libs/openzap/conf/openzap.conf > > [span wanpipe1] > number => 1 > trunk_type => e1 > b-channel => 1:1-15 > d-channel => 1:16 > b-channel => 1:17-31 > > How can i resolve the above problem. how can i load the openzap. can > some one assist me to resolve this process. > > -- > Thanks with Regards, > N.Baskar > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saeedahmad1981 at gmail.com Tue Jun 30 02:56:23 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 30 Jun 2009 11:56:23 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <20090630091456.GA5371@jdc.jasonjgw.net> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <20090630091456.GA5371@jdc.jasonjgw.net> Message-ID: Can you give me more info with CentOS. I am more comfortable with SVN trunks, can i do the same with SVN trunks? Thanks On Tue, Jun 30, 2009 at 11:14 AM, Jason White wrote: > Saeed Ahmad wrote: > > What is the best way to update to latest version if we are already > running > > an older stable version? > > You did ask for the best way, which is to build packages for your operating > system, then use your operating system's package manager to install them > and > keep track of different versions. This way, you can be sure that the right > files are installed, that old versions are cleanly deleted (unless there's > an > error in the post-installation script, in which case it's a bug) and you > can > use the package manager to find out what files are installed and where they > reside. > > You can then upgrade or downgrade simply by installing a different version > of > the package. > > FreeSWITCH supports building Debian packages, and there is also support for > Centos and Fedora. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/2a0e1b12/attachment-0001.html From benke at inqnet.at Tue Jun 30 03:09:05 2009 From: benke at inqnet.at (Christian Benke) Date: Tue, 30 Jun 2009 12:09:05 +0200 Subject: [Freeswitch-users] Is there a group variable? Message-ID: <20090630120905.536234f4@plex> Hello! I have the following scenario: I want to check if a called extension is part of a group, or as an alternative, if it is a user in the directory. My basic intention is to find out if a 3-digit extension leads to a valid user - if it doesn't, some other action will happen. Maybe there are better ways to do this than to check for the parameters above, not sure though(Don't want to use a regexp for the available extensions though). If there is such a variable, it doesn't seem to be documented or i'm still too lost in the docs... Can you give me a pointer? Regards Christian From jason at jasonjgw.net Tue Jun 30 03:09:09 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 20:09:09 +1000 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <20090630091456.GA5371@jdc.jasonjgw.net> Message-ID: <20090630100909.GA18070@jdc.jasonjgw.net> Saeed Ahmad wrote: > Can you give me more info with CentOS. > > I am more comfortable with SVN trunks, can i do the same with SVN trunks? Yes. There is a spec file in the source tree for building packages. There should be instructions on the wiki explaining how to use it - if not, someone who is more familiar with rpm-based distributions than I am will be able to refer you to instructions on how to build an rpm package. From helmut.kuper at ewetel.de Tue Jun 30 04:29:17 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 30 Jun 2009 13:29:17 +0200 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <019601c9f92b$036c02d0$0a440870$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: <4A49F70D.1080904@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Drago, sorry for correcting you, but Snom's touch scrren phone is the 870 ... http://www.snom.com/en/products/snom-870/ cheers Helmut On 30.06.2009 04:32, Drago Totev wrote: > Actually, Snom does have a version with color LCD touch screen - model 820. > I'm not sure if it is in mass production yet. > > Regarding the "nice looking"... suit yourself :-) : > http://www.snom.com/en/products/snom-820/ > > Drago -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKSfcN4tZeNddg3dwRAlKBAJ9dDp51Ki2CmY+VxEc/YvO5++xIIACfVca8 Dpc6N3A/9cPai8ujkSosv/A= =bCzJ -----END PGP SIGNATURE----- From raffaele.p.guidi at gmail.com Tue Jun 30 04:46:34 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 30 Jun 2009 13:46:34 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: Hi, I would like to give a try to this and all other "pre" releases but being tied to windows platforms (and not having a C compiler available) I would need an MSI installer. Is there a way to add a windows build to the (pre)release process (without doubling your work, of course ;)? I think one of the selling points of FreeSWITCH is the ability to seamlessly run on windows - something that asterisk cannot even dream of, and that yate promises but fails to completely fulfill. Regards, Raffaele On Tue, Jun 30, 2009 at 01:13, Michael Collins wrote: > The FreeSWITCH development team is please to announce that there's a new > version of FreeSWITCH available. Please update as soon as you reasonable > can. More details available here: > http://www.freeswitch.org/node/195 > > We appreciate everyone's help in making FreeSWITCH better. Please keep > testing and reporting back! > > -Michael > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/6f09a98a/attachment.html From jay.fenton at howlertech.com Tue Jun 30 00:03:34 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Tue, 30 Jun 2009 08:03:34 +0100 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24266762.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> Message-ID: Hi Edmar, > I need actually G729 license if there's any for freeswitch to call > mobile > phones... I already load it but has an issue on it passthrough > mode... If i > install mod_dandhi_codec it overwrites the existing G729 without > license to > a new G729 with license? Are you sure your mobile phones support G729? Most phones have hardware acceleration for AMR-NB and not G729, so most of the VoIP-over-WiFi providers use that codec instead. I believe G729 would have to be implemented in software only (without hardware assistance) and would drain the battery fairly quickly as a result. As other have pointed out on the list, and just in case you missed it mod_dahdi_codec is for use with a PCI accelerator card (the TC400B) that Digium (http://www.digium.com/) sell and will not work without it. With that card you can get up to 120 concurrent G729A-G711 calls. These days you can get much better performance from pure software codecs running on Intel/AMD systems than using the above card. We (Howler) happen to have announced a G729A codec for FreeSWITCH a few days ago - you can get more info here: http://www.howlertech.com/products/howlets/ Hope that helps. -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From jay.fenton at howlertech.com Tue Jun 30 00:11:11 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Tue, 30 Jun 2009 08:11:11 +0100 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906292020u3528f483v1462e049877c664d@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> <8cc991dd0906292020u3528f483v1462e049877c664d@mail.gmail.com> Message-ID: <9D287080-0FDE-4DD6-871D-0D00DA132438@howlertech.com> > Well the 8XX series is much prettier than the 3XX series. Though I > wonder if anyone has thought of porting FreeSWITCH to the Palm Pre? Palm Pre is Linux-based, so shouldn't be that difficult so long as you can get a working portaudio up and running (it looks like pulseaudio has already been ported, so I presume ALSA is ready and waiting in the kernel - FS might just build out of the box!). -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From lubimov at neolant.ru Tue Jun 30 07:21:06 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Tue, 30 Jun 2009 18:21:06 +0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? Message-ID: <4A4A1F52.2000307@neolant.ru> I sofia_reg.c:1765 have two user records - good #110 and bad #111. bad.xml: good.xml: Good user 110 work without any problem. But user "Bad" user 111 can't register to freeswitch. In log I can see only one message - 2009-06-30 16:31:36.590970 [WARNING] sofia_reg.c:1765 Cant register a pointer. good user exists: freeswitch at internal> user_exists id 110 neolant.ru true and bad user exists! freeswitch at internal> user_exists id 111 neolant.ru true good user don't have attr type: freeswitch at internal> user_data 110 at neolant.ru attr type -ERR no reply but bad user have attr type! : freeswitch at internal> user_data 111 at neolant.ru attr type pointer Good user have password: freeswitch at internal> user_data 110 at neolant.ru param password 123456 But bad user no have param pasword! freeswitch at internal> user_data 111 at neolant.ru param password -ERR no reply What's wrong in these configuration? How I can debug and resolve these problems? From msc at freeswitch.org Tue Jun 30 08:02:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 30 Jun 2009 08:02:55 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: <8DE8B0BC-0519-426A-A8A1-99ABDA65B930@freeswitch.org> If you're already on trunk then just do "make current" -MC Sent from my iPhone On Jun 30, 2009, at 2:04 AM, Saeed Ahmad wrote: > Hi, > > What is the best way to update to latest version if we are already > running an older stable version? > > I am using SVN trunk, sources are in /usr/src and its installed in / > usr/local/freeswitch > > >From the same src directory, is it possible to install latest rev > in somewhere else (for example /opt/freeswitch), if everything is > working ok then i replace it in /usr/local/freeswitch? > > or any other alternatives? > > Many thanks. > > > > On Tue, Jun 30, 2009 at 1:13 AM, Michael Collins > wrote: > The FreeSWITCH development team is please to announce that there's a > new version of FreeSWITCH available. Please update as soon as you > reasonable can. More details available here: > http://www.freeswitch.org/node/195 > > We appreciate everyone's help in making FreeSWITCH better. Please > keep testing and reporting back! > > -Michael > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/e36bb83c/attachment-0001.html From carlos.talbot at gmail.com Tue Jun 30 08:29:58 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Tue, 30 Jun 2009 10:29:58 -0500 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> I do make an effort to update the svn MSI every time a new release is announced. The current MSI was posted this morning (svn 14043) and should be synced up by this evening (CST). Carlos http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries On Tue, Jun 30, 2009 at 6:46 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > Hi, I would like to give a try to this and all other "pre" releases but > being tied to windows platforms (and not having a C compiler available) I > would need an MSI installer. Is there a way to add a windows build to the > (pre)release process (without doubling your work, of course ;)? I think one > of the selling points of FreeSWITCH is the ability to seamlessly run on > windows - something that asterisk cannot even dream of, and that yate > promises but fails to completely fulfill. > Regards, > Raffaele > > On Tue, Jun 30, 2009 at 01:13, Michael Collins wrote: > >> The FreeSWITCH development team is please to announce that there's a new >> version of FreeSWITCH available. Please update as soon as you reasonable >> can. More details available here: >> http://www.freeswitch.org/node/195 >> >> We appreciate everyone's help in making FreeSWITCH better. Please keep >> testing and reporting back! >> >> -Michael >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/1291e3a4/attachment.html From mike at jerris.com Tue Jun 30 08:49:30 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 11:49:30 -0400 Subject: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h In-Reply-To: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> References: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> Message-ID: <5CE0CED3-A08D-4D18-826C-7C64E8C7E1BB@jerris.com> I think that detection is not working on mac right due to it looking in default search paths. I am in process of fixing this to use in tree libtiff soon so this should fix this issue. Mike On Jun 29, 2009, at 11:08 PM, seven wrote: > Hi, > > I'm on the latest svn 14041, and I have tiff installed with "port > install tiff", how can I tell FS to find libtiff at /opt/local/ > include? > > on Linux the package should be libtiff-dev of libtiff-devel, but I > think tiff is the equivalent on Mac. > > previous versions build ok on my Mac, possible to include the libtiff > in trunk like other libs? > > checking tiffio.h usability... no > checking tiffio.h presence... no > checking for tiffio.h... no > checking pthread.h usability... yes > checking pthread.h presence... yes > checking for pthread.h... yes > checking X11/X.h usability... yes > checking X11/X.h presence... yes > checking for X11/X.h... yes > checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h > usability... yes > checking libxml/xmlmemory.h presence... yes > checking for libxml/xmlmemory.h... yes > checking libxml/parser.h usability... yes > checking libxml/parser.h presence... yes > checking for libxml/parser.h... yes > checking libxml/xinclude.h usability... yes > checking libxml/xinclude.h presence... yes > checking for libxml/xinclude.h... yes > checking FL/Fl.H usability... no > checking FL/Fl.H presence... no > checking for FL/Fl.H... no > checking FL/Fl_Overlay_Window.H usability... no > checking FL/Fl_Overlay_Window.H presence... no > checking for FL/Fl_Overlay_Window.H... no > checking FL/Fl_Light_Button.H usability... no > checking FL/Fl_Light_Button.H presence... no > checking for FL/Fl_Light_Button.H... no > checking FL/fl_draw.H usability... no > checking FL/fl_draw.H presence... no > checking for FL/fl_draw.H... no > checking FL/Fl_Cartesian.H usability... no > checking FL/Fl_Cartesian.H presence... no > checking for FL/Fl_Cartesian.H... no > checking FL/Fl_Audio_Meter.H usability... no > checking FL/Fl_Audio_Meter.H presence... no > checking for FL/Fl_Audio_Meter.H... no > checking for cos in -lm... yes > checking for library containing sinf... none required > checking for library containing cosf... none required > checking for library containing tanf... none required > checking for library containing asinf... none required > checking for library containing acosf... none required > checking for library containing atanf... none required > checking for library containing atan2f... none required > checking for library containing ceilf... none required > checking for library containing floorf... none required > checking for library containing powf... none required > checking for library containing expf... none required > checking for library containing logf... none required > checking for library containing log10f... none required > checking for TIFFOpen in -ltiff... no > configure: error: "Can't build without libtiff (does your system > require a libtiff-devel package?)" > configure: error: ./configure.gnu failed for libs/spandsp > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 30 09:03:49 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:03:49 -0400 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Message-ID: the bridge app already does all this for you doesn't it (along with bind_meta) ? Mike On Jun 27, 2009, at 2:45 AM, Dome Charoenyost wrote: > Dear All, > > I try > > s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); > if (s.ready()){ > s.setVariable("nibble_rate", "2.5"); > s.setVariable("nibble_account", "0838833133"); > s.execute("nibblebill", "heartbeat 5"); > bridge(session,s); > }; > > my question is > 1. How to cancel create s session (by dtmf ) like a * in bridge app > 2. when i hangup before s session ready is posible to cancel ? > > Best Regards. > > Dome C. From dujinfang at gmail.com Tue Jun 30 09:04:48 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 1 Jul 2009 00:04:48 +0800 Subject: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h In-Reply-To: <5CE0CED3-A08D-4D18-826C-7C64E8C7E1BB@jerris.com> References: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> <5CE0CED3-A08D-4D18-826C-7C64E8C7E1BB@jerris.com> Message-ID: Thanks. I installed libtiff from source to /usr/local/ and now it works. On Jun 30, 2009, at 11:49 PM, Michael Jerris wrote: > I think that detection is not working on mac right due to it looking > in default search paths. I am in process of fixing this to use in > tree libtiff soon so this should fix this issue. > > Mike > > On Jun 29, 2009, at 11:08 PM, seven wrote: > >> Hi, >> >> I'm on the latest svn 14041, and I have tiff installed with "port >> install tiff", how can I tell FS to find libtiff at /opt/local/ >> include? >> >> on Linux the package should be libtiff-dev of libtiff-devel, but I >> think tiff is the equivalent on Mac. >> >> previous versions build ok on my Mac, possible to include the libtiff >> in trunk like other libs? >> >> checking tiffio.h usability... no >> checking tiffio.h presence... no >> checking for tiffio.h... no >> checking pthread.h usability... yes >> checking pthread.h presence... yes >> checking for pthread.h... yes >> checking X11/X.h usability... yes >> checking X11/X.h presence... yes >> checking for X11/X.h... yes >> checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h >> usability... yes >> checking libxml/xmlmemory.h presence... yes >> checking for libxml/xmlmemory.h... yes >> checking libxml/parser.h usability... yes >> checking libxml/parser.h presence... yes >> checking for libxml/parser.h... yes >> checking libxml/xinclude.h usability... yes >> checking libxml/xinclude.h presence... yes >> checking for libxml/xinclude.h... yes >> checking FL/Fl.H usability... no >> checking FL/Fl.H presence... no >> checking for FL/Fl.H... no >> checking FL/Fl_Overlay_Window.H usability... no >> checking FL/Fl_Overlay_Window.H presence... no >> checking for FL/Fl_Overlay_Window.H... no >> checking FL/Fl_Light_Button.H usability... no >> checking FL/Fl_Light_Button.H presence... no >> checking for FL/Fl_Light_Button.H... no >> checking FL/fl_draw.H usability... no >> checking FL/fl_draw.H presence... no >> checking for FL/fl_draw.H... no >> checking FL/Fl_Cartesian.H usability... no >> checking FL/Fl_Cartesian.H presence... no >> checking for FL/Fl_Cartesian.H... no >> checking FL/Fl_Audio_Meter.H usability... no >> checking FL/Fl_Audio_Meter.H presence... no >> checking for FL/Fl_Audio_Meter.H... no >> checking for cos in -lm... yes >> checking for library containing sinf... none required >> checking for library containing cosf... none required >> checking for library containing tanf... none required >> checking for library containing asinf... none required >> checking for library containing acosf... none required >> checking for library containing atanf... none required >> checking for library containing atan2f... none required >> checking for library containing ceilf... none required >> checking for library containing floorf... none required >> checking for library containing powf... none required >> checking for library containing expf... none required >> checking for library containing logf... none required >> checking for library containing log10f... none required >> checking for TIFFOpen in -ltiff... no >> configure: error: "Can't build without libtiff (does your system >> require a libtiff-devel package?)" >> configure: error: ./configure.gnu failed for libs/spandsp >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 30 09:05:28 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:05:28 -0400 Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? In-Reply-To: References: Message-ID: <2F02DCD0-71EF-4787-9BE6-D4D667E53132@jerris.com> If you don't need authentication, you don't need a gateway, if you do, you will need to setup a user on the other box to register to. On Jun 30, 2009, at 3:05 AM, Brad Tuan wrote: > I know that i need to set the dialplan, > > my problem is when FS_B send a REGISTER to FS_A, FS_A will return a > 403 to FS_B > > Like this: > > 2009-06-30 15:03:25 [NOTICE] sofia_reg.c:305 > sofia_reg_check_gateway() Registering FS_A > 2009-06-30 15:03:25 [ERR] sofia_reg.c:1391 > sofia_reg_handle_sip_r_register() > FS_A Registration Failed with status Forbidden > [403]. failure #2 > 2009-06-30 15:03:25 [WARNING] sofia_reg.c:334 > sofia_reg_check_gateway() > FS_A Failed Registration, setting retry to 30 > seconds. > > > Create a dialplan for FS A to communicate to FS B. > > > > dialplan/default/00_fsa.xml > > > > > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > > > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > > > data="sofia/gateway/FreeSwitch/1$218.210.xxx.xxx"/> -> IP of > FS B > > > > > > > To check if your gateway succeed > > > just type on sofia status check if gateway FreeSWITCH is there > connection is > > set and ready to go :) From mike at jerris.com Tue Jun 30 09:10:20 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:10:20 -0400 Subject: [Freeswitch-users] Is there a group variable? In-Reply-To: <20090630120905.536234f4@plex> References: <20090630120905.536234f4@plex> Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#in_group http://wiki.freeswitch.org/wiki/Mod_commands#user_exists On Jun 30, 2009, at 6:09 AM, Christian Benke wrote: > Hello! > > I have the following scenario: > > I want to check if a called extension is part of a group, or as an > alternative, if it is a user in the directory. > My basic intention is to find out if a 3-digit extension leads to a > valid user - if it doesn't, some other action will happen. > Maybe there are better ways to do this than to check for the > parameters > above, not sure though(Don't want to use a regexp for the available > extensions though). > > If there is such a variable, it doesn't seem to be documented or i'm > still too lost in the docs... Can you give me a pointer? From mike at jerris.com Tue Jun 30 09:12:26 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:12:26 -0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? In-Reply-To: <4A4A1F52.2000307@neolant.ru> References: <4A4A1F52.2000307@neolant.ru> Message-ID: you have a pointer somewhere in your directory for that user, hard to see without seeing the whole config, but grep for 111 and see what else you find. Mike On Jun 30, 2009, at 10:21 AM, Alexey Lubimov wrote: > I sofia_reg.c:1765 have two user records - good #110 and bad #111. > > bad.xml: > > > > > > > > > value="domestic,international,local"/> > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > good.xml: > > > > > > > > > value="domestic,international,local"/> > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > Good user 110 work without any problem. But user "Bad" user 111 can't > register to freeswitch. > > In log I can see only one message - 2009-06-30 16:31:36.590970 > [WARNING] sofia_reg.c:1765 Cant register a pointer. > > good user exists: > freeswitch at internal> user_exists id 110 neolant.ru > true > > and bad user exists! > freeswitch at internal> user_exists id 111 neolant.ru > true > > good user don't have attr type: > > freeswitch at internal> user_data 110 at neolant.ru attr type > -ERR no reply > > but bad user have attr type! : > > freeswitch at internal> user_data 111 at neolant.ru attr type > pointer > > > Good user have password: > > freeswitch at internal> user_data 110 at neolant.ru param password > 123456 > > But bad user no have param pasword! > > freeswitch at internal> user_data 111 at neolant.ru param password > -ERR no reply > > > What's wrong in these configuration? How I can debug and resolve these > problems? From mike at jerris.com Tue Jun 30 09:07:13 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:07:13 -0400 Subject: [Freeswitch-users] Any advances on T.38 support for FS? In-Reply-To: <1246353349.30167.83.camel@luna.tc.commsmundi.com> References: <1246353349.30167.83.camel@luna.tc.commsmundi.com> Message-ID: <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> We currently support t.38 passthrough only using proxy_media mode. T. 38 gateway is on the roadmap but not yet close to complete. Mike On Jun 30, 2009, at 5:15 AM, Fran?ois Delawarde wrote: > Many issues on Asterisk's T.38 (or probably just on T.38?)... > > Could it convince those relying on this "modern" version of a 50yo > technology to switch to and with FreeSwitch? From msc at freeswitch.org Tue Jun 30 09:50:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 30 Jun 2009 09:50:29 -0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Message-ID: <87f2f3b90906300950m6514972dxa26c43bfbb9785ce@mail.gmail.com> On Tue, Jun 30, 2009 at 9:03 AM, Michael Jerris wrote: > the bridge app already does all this for you doesn't it (along with > bind_meta) ? > > Mike > In other words, everything you want is available in the dialplan with no overheard from launching a JS. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/cf260881/attachment.html From raffaele.p.guidi at gmail.com Tue Jun 30 10:45:29 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 30 Jun 2009 19:45:29 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> Message-ID: Ok, so tomorrow I'll find http://files.freeswitch.org/freeswitch-1.0.4pre9.msi? Thanks a lot, Raffaele On Tue, Jun 30, 2009 at 17:29, Carlos Talbot wrote: > I do make an effort to update the svn MSI every time a new release is > announced. The current MSI was posted this morning (svn 14043) and should be > synced up by this evening (CST). > Carlos > > > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries > > > On Tue, Jun 30, 2009 at 6:46 AM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > >> Hi, I would like to give a try to this and all other "pre" releases but >> being tied to windows platforms (and not having a C compiler available) I >> would need an MSI installer. Is there a way to add a windows build to the >> (pre)release process (without doubling your work, of course ;)? I think one >> of the selling points of FreeSWITCH is the ability to seamlessly run on >> windows - something that asterisk cannot even dream of, and that yate >> promises but fails to completely fulfill. >> Regards, >> Raffaele >> >> On Tue, Jun 30, 2009 at 01:13, Michael Collins wrote: >> >>> The FreeSWITCH development team is please to announce that there's a new >>> version of FreeSWITCH available. Please update as soon as you reasonable >>> can. More details available here: >>> http://www.freeswitch.org/node/195 >>> >>> We appreciate everyone's help in making FreeSWITCH better. Please keep >>> testing and reporting back! >>> >>> -Michael >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/66bc7fc1/attachment.html From brian at freeswitch.org Tue Jun 30 11:01:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 30 Jun 2009 13:01:32 -0500 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> Message-ID: <223FE753-8F58-4196-B9A1-8F9BE8D08067@freeswitch.org> Maybe! We might hold it hostage! ;) muahhahaha /b On Jun 30, 2009, at 12:45 PM, Raffaele P. Guidi wrote: > Ok, so tomorrow I'll find http://files.freeswitch.org/freeswitch-1.0.4pre9.msi? > > Thanks a lot, > Raffaele -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/6d8d446d/attachment.html From marketing at cluecon.com Tue Jun 30 14:58:50 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 30 Jun 2009 14:58:50 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Great News: Early Bird Extended Message-ID: <87f2f3b90906301458n584ea8e8ta848cf1f184a8976@mail.gmail.com> Great news for all of those who've not been able to sign up yet for ClueCon: we've decided to extend the early bird special through Tuesday July 21! Note: this is also the last day you can book a room with the Wyndham, but please don't wait until the last minute. More good news and reasons to sign up right away: fun giveaways! Our sponsors and partners are supplying goodies to give away to conference attendees. Here are some highlights: >From Sangoma - B600 cards (qty 3) and A101D (qty 1) >From Snom - 20 phones, mix of models 320, 360, 270 Stay tuned for more updates and surprises about ClueCon 2009! -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/f80f31ee/attachment.html From darklion11 at yahoo.com Tue Jun 30 18:46:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 30 Jun 2009 18:46:01 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> Message-ID: <24282895.post@talk.nabble.com> So what codec supports mobile phones? jfenton wrote: > > > Hi Edmar, > >> I need actually G729 license if there's any for freeswitch to call >> mobile >> phones... I already load it but has an issue on it passthrough >> mode... If i >> install mod_dandhi_codec it overwrites the existing G729 without >> license to >> a new G729 with license? > > Are you sure your mobile phones support G729? Most phones have hardware > acceleration for AMR-NB and not G729, so most of the VoIP-over-WiFi > providers use that codec instead. I believe G729 would have to be > implemented in software only (without hardware assistance) and would > drain the battery fairly quickly as a result. > > As other have pointed out on the list, and just in case you missed it > mod_dahdi_codec is for use with a PCI accelerator card (the TC400B) > that Digium (http://www.digium.com/) sell and will not work without > it. With that card you can get up to 120 concurrent G729A-G711 calls. > > These days you can get much better performance from pure software > codecs running on Intel/AMD systems than using the above card. We > (Howler) happen to have announced a G729A codec for FreeSWITCH a > few days ago - you can get more info here: > > http://www.howlertech.com/products/howlets/ > > Hope that helps. > > -- > Regards, > > Jay Fenton, CTO > Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ > tel: +44 207 099 7095 fax: +44 207 099 7098 > http://www.howlertech.com/ > http://www.linkedin.com/in/jfenton > > Registered in England & Wales, Company No. 06285634 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24282895.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From craig at overthewire.com.au Tue Jun 30 19:03:38 2009 From: craig at overthewire.com.au (Craig Askings) Date: Wed, 1 Jul 2009 12:03:38 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24282895.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <8cc991dd0906301903k2dd15de9tdd9b7b9a77338ce1@mail.gmail.com> GSM 2009/7/1 Edmar Cruz : > > So what codec supports mobile phones? > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From steveu at coppice.org Tue Jun 30 20:05:20 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 01 Jul 2009 11:05:20 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24282895.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <4A4AD270.30204@coppice.org> Edmar Cruz wrote: > So what codec supports mobile phones? > The main codecs used by mobile phones are: GSM FR The original GSM code, largely replaced by later codecs (some VoIP stuff uses this) GSM HR The half rate codec for GSM GSM EFR A later improved full rate codec that largely replaced GSM FR on GSM networks AMR-NB The current start of the art narrowband codec for GSM and UMTS networks AMR-WB The wide band codec for UMTS, though most networks seem to block it EVRC The main codec for the CDMA networks Each of these codecs sounds pretty respectable, and G.729 sounds pretty respectable. However, if you transcode from one to another the result can be pretty bad. Steve From rupa at rupa.com Tue Jun 30 20:22:27 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 30 Jun 2009 22:22:27 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A4AD270.30204@coppice.org> References: <24251951.post@talk.nabble.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <4A4AD270.30204@coppice.org> Message-ID: And unless you are directly connecting to the cell phone provider you are going to be converted to ULAW/ALAW to traverse the PSTN. So there is no advantage going to a native GSM codec only to have it expanded out to G711 and then back to GSM codec. On Tue, Jun 30, 2009 at 10:05 PM, Steve Underwood wrote: > Edmar Cruz wrote: > > So what codec supports mobile phones? > > > The main codecs used by mobile phones are: > GSM FR The original GSM code, largely replaced by later > codecs (some VoIP stuff uses this) > GSM HR The half rate codec for GSM > GSM EFR A later improved full rate codec that largely replaced > GSM FR on GSM networks > AMR-NB The current start of the art narrowband codec for GSM > and UMTS networks > AMR-WB The wide band codec for UMTS, though most networks > seem to block it > EVRC The main codec for the CDMA networks > > Each of these codecs sounds pretty respectable, and G.729 sounds pretty > respectable. However, if you transcode from one to another the result > can be pretty bad. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/33157112/attachment.html From darklion11 at yahoo.com Tue Jun 30 20:43:34 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 30 Jun 2009 20:43:34 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A4AD270.30204@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <4A4AD270.30204@coppice.org> Message-ID: <24283663.post@talk.nabble.com> I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i shall load to G729 for freeswitch that needs a license? Steve Underwood wrote: > > Edmar Cruz wrote: >> So what codec supports mobile phones? >> > The main codecs used by mobile phones are: > GSM FR The original GSM code, largely replaced by later > codecs (some VoIP stuff uses this) > GSM HR The half rate codec for GSM > GSM EFR A later improved full rate codec that largely replaced > GSM FR on GSM networks > AMR-NB The current start of the art narrowband codec for GSM > and UMTS networks > AMR-WB The wide band codec for UMTS, though most networks > seem to block it > EVRC The main codec for the CDMA networks > > Each of these codecs sounds pretty respectable, and G.729 sounds pretty > respectable. However, if you transcode from one to another the result > can be pretty bad. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24283663.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mythicalbox at weavver.com Tue Jun 30 21:11:36 2009 From: mythicalbox at weavver.com (Mitchel Constantin) Date: Tue, 30 Jun 2009 21:11:36 -0700 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? Message-ID: Hello, I'm experiencing a bug that I've been working on most of today. I can not call between two SIP phones that register successfully. In order to diagnose it, I have removed my FreeSWITCH server out of the NAT/firewall to try and eliminate any such issues with these things. Here is how I ran into the issue: 1. Started from sample configs 2. Enabled xml_curl and wrote the associated script to generate XML and had the phones authenticating but was forced to use the IP address of the server as the domain in my directory XML. 3. I tested calls and they worked using this syntax: originate sofia/internal/mythicalbox%205.134.225.20 (the server's ip) sofia/internal/johndoe%205.134.225.20 4. Next to remove the "limit" on using only the IP as the domain for users, I commented out force-register-domain and force-register-db-domain in internal.xml. 5. My phones now register using the correct domain name (i.e. weavver.com) instead of the IP address (205.134.225.20) as the domain. 6. Now the problem... My originate command no longer works using the new syntax: originate sofia/internal/mythicalbox%weavver.comsofia/internal/johndoe% weavver.com The phones do show up as registered when I type "sofia status profile internal": API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 205.134.225.20 SIP-IP 205.134.225.20 URL sip:mod_sofia at 205.134.225.20:5060 BIND-URL sip:mod_sofia at 205.134.225.20:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 9 FAILED-CALLS-IN 3 CALLS-OUT 8 FAILED-CALLS-OUT 18 Registrations: ================================================================================================= Call-ID: ZWQ3NjRhZWI3MDc4ZDhjNTdhZDU2ZGVkM2JmYjg3NTc. User: mythicalbox at weavver.com Contact: "mythicalbox" Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-06-30 22:43:47) Host: duck.weavver.com IP: 64.183.110.250 Port: 65438 Auth-User: mythicalbox Auth-Realm: weavver.com Call-ID: Y2E0NWZiZWZjOTYxYjVhNmQ5ZDkyMzJjMzUxYWM0ZGM. User: johndoe at weavver.com Contact: "johndoe" Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-06-30 22:47:37) Host: duck.weavver.com IP: 64.183.110.250 Port: 65512 Auth-User: johndoe Auth-Realm: weavver.com ================================================================================================= FYI, FreeSWITCH is on a public IP address on the 'nets and the phones are behind the same firewall on a different public IP address on the internet. Thank you in advance for any help! :) -- Mitchel Constantin Weavver, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/43fe93c4/attachment.html From msc at freeswitch.org Tue Jun 30 21:40:12 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 30 Jun 2009 21:40:12 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24283663.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <4A4AD270.30204@coppice.org> <24283663.post@talk.nabble.com> Message-ID: <39D35AD1-05A2-4D42-AEF1-09216777B66C@freeswitch.org> On Jun 30, 2009, at 8:43 PM, Edmar Cruz wrote: > > I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i > shall load > to G729 for freeswitch that needs a license? > You need a license if you are transcoding to or from G729. If you are just passing the media stream or you stay out of the media stream then you do not need a license. -MC > Steve Underwood wrote: >> >> Edmar Cruz wrote: >>> So what codec supports mobile phones? >>> >> The main codecs used by mobile phones are: >> GSM FR The original GSM code, largely replaced by later >> codecs (some VoIP stuff uses this) >> GSM HR The half rate codec for GSM >> GSM EFR A later improved full rate codec that largely >> replaced >> GSM FR on GSM networks >> AMR-NB The current start of the art narrowband codec for >> GSM >> and UMTS networks >> AMR-WB The wide band codec for UMTS, though most networks >> seem to block it >> EVRC The main codec for the CDMA networks >> >> Each of these codecs sounds pretty respectable, and G.729 sounds >> pretty >> respectable. However, if you transcode from one to another the result >> can be pretty bad. >> >> Steve >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24283663.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yudha2008 at gmail.com Tue Jun 30 23:29:14 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 1 Jul 2009 11:59:14 +0530 Subject: [Freeswitch-users] Error in OpenZap In-Reply-To: <1246354661.19445.13.camel@raul-laptop> References: <1246354661.19445.13.camel@raul-laptop> Message-ID: *Hi, i have changed the openzap.conf file but still i get the same error **[span wanpipe 1] number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31* * freeswitch at localhost.localdomain> load mod_libpri API CALL [load(mod_libpri)] output: -ERR [module load file routine returned an error] 2009-07-01 11:27:51 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_libpri.so **/usr/local/freeswitch/mod/mod_libpri.so: cannot open shared object file: No such file or directory** freeswitch at localhost.localdomain> load mod_openzap 2009-07-01 11:28:04 [NOTICE] zap_io.c:2626 zap_global_init() Modules configured: 1 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '1' API CALL [load(mod_openzap)] output: -ERR [module load file routine returned an error] 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:1-15' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '1:16' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:17-31' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '2' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:1-15' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:17-31' 2009-07-01 11:28:04 [INFO] zap_io.c:2370 load_config() Configured 0 channel(s) 2009-07-01 11:28:04 [ERR] zap_io.c:2633 zap_global_init() No modules configured! 2009-07-01 11:28:04 [ERR] mod_openzap.c:2401 mod_openzap_load() Error loading OpenZAP 2009-07-01 11:28:04 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so **Module load routine returned an error** * *-- Thanks with Regards, N.Baskar* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/bcc735c4/attachment-0001.html From god.nirvana at gmail.com Tue Jun 30 23:48:03 2009 From: god.nirvana at gmail.com (qian ma) Date: Wed, 1 Jul 2009 14:48:03 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? Message-ID: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> hi all freeswitch support PCMU only? i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but freeswitch still support PCMU only, below is the trace: 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9876 at 58.212.219.104 [KILL] 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal sofia/maq/9876 at 58.212.219.104 [BREAK] 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( sofia/maq/9876 at 58.212.219.104) State HANGUP how to configure the freeswitch?? support more codecs??? thx! m.q -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/c3ebec91/attachment.html From solko at gcdf.pl Mon Jun 1 01:21:34 2009 From: solko at gcdf.pl (Szymon Olko) Date: Mon, 01 Jun 2009 10:21:34 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <200906010938513832282@163.com> References: <200906010938513832282@163.com> Message-ID: <4A238F8E.8060400@gcdf.pl> zhaoxxqq pisze: > HI, > I use event socket to send command to FS conference. > I send " conference testconf play /root/test.wav" in console. It worked ok. > I send "api conference testconf play /record/test.wav" by event socket. > and the response is"Disconneted, Good bye.See you at ClueCon..". > I changed the wav file to www root. the same problem. can you help me? > 2009-06-01 Do you use 'auth ClueCon' before sending 'api' command? Szymon From codecomplete at free.fr Mon Jun 1 02:11:13 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 1 Jun 2009 02:11:13 -0700 (PDT) Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <1243809075.31679.2.camel@sodium> References: <23807353.post@talk.nabble.com> <1243809075.31679.2.camel@sodium> Message-ID: <23811643.post@talk.nabble.com> Hadley Rich wrote: > The issue you are going to have is that the IP0x are based on the > blackfin processor which as far as I'm aware FreeSWITCH doesn't compile on > yet. Last I heard there's an issue with APR. Too bad. Thanks for the tip. -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p23811643.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 1 02:57:07 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 11:57:07 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. /Peter -----Ursprungligt meddelande----- Fr?n: Peter Olsson Skickat: den 30 maj 2009 09:01 Till: freeswitch-users at lists.freeswitch.org ?mne: FW: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? And just to be clear, even though media flows in one direction (from FS to phone), I get no audio at all. And by the way, I mean SVN, not SNV :) Sorry for double posting... /Peter ________________________________________ Fr?n: Peter Olsson Skickat: den 30 maj 2009 08:58 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Michael Jerris [mike at jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I'm on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: Nope - it's not :) Just to make sure I even deleted the source completely, and checked everything out again. Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682! From anthony.minessale at gmail.com Mon Jun 1 05:59:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 07:59:30 -0500 Subject: [Freeswitch-users] Custom variable and channel answer event (socket) In-Reply-To: <4A232666.70000@gmail.com> References: <4A232666.70000@gmail.com> Message-ID: <191c3a030906010559j5b32ffd6j7c329c9283dcd9df@mail.gmail.com> all the variables should be present in that event. Maybe rebuild to make sure you don't have skew from an upgrade. On Sun, May 31, 2009 at 7:52 PM, paul.degt at gmail.com wrote: > Hi, > I am setting a custom var from a javascript code, I do see it in channel > state events and others up to channel answer. In channel answer event it > somehow disappears, > and then comes back in channel destroy event. My problem is that I > really need it in channel answer event. > What can be wrong here? I did put verbose_event action everywhere I > could think of. > Help would be greatly appreciated. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/a0ad47f9/attachment-0002.html From steveu at coppice.org Mon Jun 1 06:02:22 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 01 Jun 2009 21:02:22 +0800 Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <23807353.post@talk.nabble.com> References: <23807353.post@talk.nabble.com> Message-ID: <4A23D15E.3040908@coppice.org> Fred-145 wrote: > Hello > > Atcom's IP01 unit (www.atcom.cn) can be expanded to have 128MB RAM and 1GB > NAND flash. Before I go ahead and check, would someone know if a minimal > Linux + LAMP server* + Freeswitch can run OK with this amount of memory? > > Thank you. > > * I think I'll trade MySQL with Firebird, to avoid buying a license for my > commercial application > I don't know any free telephony project which will build and run on an IP01, except the ones which have specifically been adapted for the Blackfin. If you look at www.rowetel.com or www.astfin.org you will find versions of Asterisk which have been adapted for the IP01. Nobody has yet adapted Freeswitch for the Blackfin, and they probably won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux, which is a cut down Linux for machines of this type. It is quite troublesome to get memory management to behave sanely on a machine without an MMU. The Asterisk adaptions for the Blackfin have problems with this too, but if you don't let the memory become too fragmented they work OK. The lack of floating point hardware in the Blackfin, and a number of other embedded processors, can also cause performance issues. The core functions of Freeswitch work reasonably well with emulated floating point, but some things, like the FAX engine, are really too slow to be very practical until more of the code is adapted to provide a fixed point version. Steve From anthony.minessale at gmail.com Mon Jun 1 06:08:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 08:08:12 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A21E87E.90709@gmail.com> References: <4A1BFECE.7070603@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> Message-ID: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> I added support so when multiple endconf users are in the same conference the total number of people with the flag must reach 0 before it kills the conf. Can I take a small break now please ;) That's why I'm afraid to add new stuff sometimes. On Sat, May 30, 2009 at 9:16 PM, wrote: > I think I can answer my own question after looking at the code... It > seems that when THAT ONE user leaves, a flag is set that notifies the > conference thread to teardown the conference. I guess I will have to > roll my own on this one I guess, especially since I don't want to kill > the conference completely, just drop the users back to music. > > Also, more importantly... > I just discovered a number of conference profile options that are > neither documented in the Wiki nor mentioned in the sample configuration > file. I've added entries in the Wiki for all the ones that were > missing, but I don't know what half of them do. =( > > Could someone in-the-know please fill those in? Also, I would suggest > adding those to the sample config file. Options like "endconf" and > "announce-user" are GREAT conference features, but no one knows they are > there! > > (I had actually implemented the user count announcement within > Javascript, because I didn't know it was available.) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/351dcf77/attachment-0002.html From kristian.kielhofner at gmail.com Mon Jun 1 06:19:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 1 Jun 2009 09:19:36 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> Message-ID: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson wrote: > Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. > > /Peter > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From larclap at yahoo.com Mon Jun 1 06:20:17 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 1 Jun 2009 06:20:17 -0700 Subject: [Freeswitch-users] Rotating log files not working In-Reply-To: <5a8712120905311837r53efb1d0j3a32ad3661e3bfdf@mail.gmail.com> References: <001801c9e216$287e1800$797a4800$@com> <5a8712120905311837r53efb1d0j3a32ad3661e3bfdf@mail.gmail.com> Message-ID: <004201c9e2bb$b5064ae0$1f12e0a0$@com> I am using version 13441 on Centos 5. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Sunday, May 31, 2009 6:38 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Rotating log files not working Just for the record, always update do latest trunk when testing and provide revision number (version command). Later, jmesquita On Sun, May 31, 2009 at 2:35 PM, Lars Zeb wrote: I am trying to rotate the logs, specifically the cdr ones. But the existing extension and Master csv files are not rotated; they remain untouched. I issue the command ?kill ?s HUP pid? (pid of freeswitch). The fs console says 2009-05-31 10:25:58 [NOTICE] mod_logfile_c:157 mod_logfile_rotate() New log started. The conf/autoload_configs/cdr-csv.conf.xml shows: What am I doing wrong here? Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/32a271ea/attachment-0002.html From peter.olsson at visionutveckling.se Mon Jun 1 06:32:38 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 15:32:38 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Thanks for the reply! Will this really help though? From what I understand of the change that breaks the compatibility, it will always send this header in the "200 OK" message (my problem is incoming calls to FS). The fix you're describing, isn't it when calling from FS to the other end? This way it works either way, it's just when the PBX gets this in the 200 OK message with this header that it stops working. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Kristian Kielhofner Skickat: den 1 juni 2009 15:20 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson wrote: > Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. > > /Peter > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a23d69e32932135138353! From brian at freeswitch.org Mon Jun 1 06:32:59 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:32:59 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> No he's talking about the one from SFSIP-111, Again that shouldn't matter... I'll fix that. /b On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote: > Peter, > > Ouch. Your PBX is broken. It shouldn't do that. > > Luckily FreeSWITCH provides a way to select RPID/PAI/none: > > http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From brian at freeswitch.org Mon Jun 1 06:36:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:36:51 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Message-ID: I'll fix it where it won't do it on anything but polycom. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: > Thanks for the reply! > > Will this really help though? From what I understand of the change > that breaks the compatibility, it will always send this header in > the "200 OK" message (my problem is incoming calls to FS). The fix > you're describing, isn't it when calling from FS to the other end? > This way it works either way, it's just when the PBX gets this in > the 200 OK message with this header that it stops working. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/db9e7759/attachment-0002.html From brian at freeswitch.org Mon Jun 1 06:46:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 08:46:39 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DA7@cooper> Message-ID: <24AB9CC7-9E06-44DF-95EB-5437BFFF4324@freeswitch.org> Your PBX is broken but now I have fixed it to only do that feature with Polycom for now. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote: > Thanks for the reply! > > Will this really help though? From what I understand of the change > that breaks the compatibility, it will always send this header in > the "200 OK" message (my problem is incoming calls to FS). The fix > you're describing, isn't it when calling from FS to the other end? > This way it works either way, it's just when the PBX gets this in > the 200 OK message with this header that it stops working. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3e1f4fc1/attachment-0002.html From kristian.kielhofner at gmail.com Mon Jun 1 06:51:37 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 1 Jun 2009 09:51:37 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> Message-ID: <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Ahh... That's what I get for not reading the entire thread! On Mon, Jun 1, 2009 at 9:32 AM, Brian West wrote: > No he's talking about the one from SFSIP-111, Again that shouldn't > matter... I'll fix that. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Jun 1 07:02:26 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 09:02:26 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Message-ID: I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote: > Ahh... That's what I get for not reading the entire thread! > > On Mon, Jun 1, 2009 at 9:32 AM, Brian West > wrote: >> No he's talking about the one from SFSIP-111, Again that shouldn't >> matter... I'll fix that. >> >> /b >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From peter.olsson at visionutveckling.se Mon Jun 1 08:08:53 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 1 Jun 2009 17:08:53 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> <6C92900F-A110-4656-A48F-30066208D510@freeswitch.org> <2d9149cd0906010651t2a896996i250a289b43035e94@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57DDB@cooper> Brian - you're the man! :) /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 1 juni 2009 16:02 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote: > Ahh... That's what I get for not reading the entire thread! > > On Mon, Jun 1, 2009 at 9:32 AM, Brian West > wrote: >> No he's talking about the one from SFSIP-111, Again that shouldn't >> matter... I'll fix that. >> >> /b >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a23e16232931225569523! From brian at freeswitch.org Mon Jun 1 08:13:50 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 10:13:50 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57D3E@cooper> <2d9149cd0906010619v3bbab768n6a19c9ae822c6886@mail.gmail.com> Message-ID: <34D78532-AB64-4D28-A6DB-2586390D271A@freeswitch.org> Makes me wanna take a clue-by-4 to some of these devices and just beat them to death! :P /b On Jun 1, 2009, at 8:19 AM, Kristian Kielhofner wrote: > Peter, > > Ouch. Your PBX is broken. It shouldn't do that. > > Luckily FreeSWITCH provides a way to select RPID/PAI/none: > > http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/766c3c34/attachment-0002.html From brian at freeswitch.org Mon Jun 1 08:26:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 10:26:42 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <4A238F8E.8060400@gcdf.pl> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: Did you happen to check out the ClueCon website? Link below! :P /b On Jun 1, 2009, at 3:21 AM, Szymon Olko wrote: > Do you use 'auth ClueCon' before sending 'api' command? > > Szymon Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/832d6d00/attachment-0002.html From asannucci at gmail.com Mon Jun 1 09:39:32 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 18:39:32 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: I Love FreeSWITCH and ClueCon '09 :) The spanish comunity too. www.freeswitch.es -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/7153f99a/attachment-0002.html From brian at freeswitch.org Mon Jun 1 09:49:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 11:49:51 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: So we can look forward to seeing you at ClueCon? /b On Jun 1, 2009, at 11:39 AM, bakko wrote: > I Love FreeSWITCH and ClueCon '09 > > :) > > The spanish comunity too. > www.freeswitch.es > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/efeb9629/attachment-0002.html From fvillarroel at yahoo.com Mon Jun 1 09:54:09 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 09:54:09 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <458378.32320.qm@web34303.mail.mud.yahoo.com> Dear all. I have problem with g729 passthru mode. I received traffic from a Asterisk on my FS and forward to other Asterisk, when i use codec ulaw this works very well. But when i try use G729 i received the following messages and SIP Trace: 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/admin/42452904 at 190.208.xx.yy [f65514e0-4ec7-11de-9b78-150e2985561f] 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/42452904 at 190.208.xx.yy entering state [received][100] 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 25643 25643 IN IP4 190.208.xx.yy s=session c=IN IP4 190.208.xx.yy t=0 0 m=audio 10236 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/42452904 at 190.208.xx.yy [KILL] 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904 at 190.208.xx.yy [BREAK] send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 From: "102" ;tag=bmke36jc1v To: "102" ;tag=rZp0XXrK9NHFD Call-ID: 3c26700b249f-sryanqz0td8u at snom360-00041323143F CSeq: 27056 REGISTER Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 Date: Mon, 01 Jun 2009 16:19:20 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running State Change CS_HANGUP 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/42452904 at 190.208.xx.yy hanging up, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 488 send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 From: "452904" ;tag=as4e2616ae To: ;tag=S8FSZr9p6y71r Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/42452904 at 190.208.xx.yy Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_HANGUP -> CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/42452904 at 190.208.xx.yy [BREAK] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running State Change CS_REPORTING 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: ------------------------------------------------------------------------ ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport From: "452904" ;tag=as4e2616ae To: ;tag=S8FSZr9p6y71r Contact: Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/42452904 at 190.208.xx.yy Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING going to sleep 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_REPORTING -> CS_DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) Locked, Waiting on external entities 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) Ended 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/42452904 at 190.208.xx.yy SOFIA DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/42452904 at 190.208.xx.yy Standard DESTROY 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY going to sleep My vars.xml : I hope your comments for know where is the config problem Fernando. From asannucci at gmail.com Mon Jun 1 09:54:13 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 18:54:13 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: Maybe next year. On August i have to work :(. If you organize ClueCon 2010 on setpember I wil go. :) Good luck. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/97e09d41/attachment-0002.html From msc at freeswitch.org Mon Jun 1 09:59:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 09:59:34 -0700 Subject: [Freeswitch-users] Custom variable and channel answer event (socket) In-Reply-To: <4A232666.70000@gmail.com> References: <4A232666.70000@gmail.com> Message-ID: <87f2f3b90906010959p787a83f1nd051ea3ac460e8a3@mail.gmail.com> Can you pastebin a simple script that demonstrates the issue? Also, include any dialplan/configuration changes you use. Finally, please paste in a sample event with and without the variable in question. Once we have more information we will see what we can do to help. -MC On Sun, May 31, 2009 at 5:52 PM, paul.degt at gmail.com wrote: > Hi, > I am setting a custom var from a javascript code, I do see it in channel > state events and others up to channel answer. In channel answer event it > somehow disappears, > and then comes back in channel destroy event. My problem is that I > really need it in channel answer event. > What can be wrong here? I did put verbose_event action everywhere I > could think of. > Help would be greatly appreciated. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/a5ba4597/attachment-0002.html From stevecrozz at gmail.com Mon Jun 1 10:09:47 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 1 Jun 2009 10:09:47 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> Message-ID: <11990ade0906011009v227a0d80r5b965a48e67ee5dd@mail.gmail.com> No breaks! keep improving the conference app :) --Stephen On Mon, Jun 1, 2009 at 6:08 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I added support so when multiple endconf users are in the same conference > the total number of people with the flag must reach > 0 before it kills the conf. > > Can I take a small break now please ;) > That's why I'm afraid to add new stuff sometimes. > > > > On Sat, May 30, 2009 at 9:16 PM, wrote: > >> I think I can answer my own question after looking at the code... It >> seems that when THAT ONE user leaves, a flag is set that notifies the >> conference thread to teardown the conference. I guess I will have to >> roll my own on this one I guess, especially since I don't want to kill >> the conference completely, just drop the users back to music. >> >> Also, more importantly... >> I just discovered a number of conference profile options that are >> neither documented in the Wiki nor mentioned in the sample configuration >> file. I've added entries in the Wiki for all the ones that were >> missing, but I don't know what half of them do. =( >> >> Could someone in-the-know please fill those in? Also, I would suggest >> adding those to the sample config file. Options like "endconf" and >> "announce-user" are GREAT conference features, but no one knows they are >> there! >> >> (I had actually implemented the user count announcement within >> Javascript, because I didn't know it was available.) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/399de241/attachment-0002.html From anthony.minessale at gmail.com Mon Jun 1 10:11:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 12:11:25 -0500 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> Message-ID: <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> It's always august =D it's ok. it counts as work! On Mon, Jun 1, 2009 at 11:54 AM, bakko wrote: > Maybe next year. > > On August i have to work :(. > > If you organize ClueCon 2010 on setpember I wil go. > > :) > > Good luck. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/aa87e413/attachment-0002.html From msc at freeswitch.org Mon Jun 1 10:15:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 10:15:18 -0700 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <458378.32320.qm@web34303.mail.mud.yahoo.com> References: <458378.32320.qm@web34303.mail.mud.yahoo.com> Message-ID: <87f2f3b90906011015u16cf2c0h54829fe8f73372c0@mail.gmail.com> What does your dialplan look like? Just curious where/how you set proxy-media mode. -MC On Mon, Jun 1, 2009 at 9:54 AM, FERNANDO VILLARROEL wrote: > > Dear all. > > I have problem with g729 passthru mode. > > I received traffic from a Asterisk on my FS and forward to other Asterisk, > when i use codec ulaw this works very well. > > But when i try use G729 i received the following messages and SIP Trace: > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel sofia/admin/42452904 at 190.208.xx.yy[f65514e0-4ec7-11de-9b78-150e2985561f] > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/admin/42452904 at 190.208.xx.yy entering state [received][100] > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=root 25643 25643 IN IP4 190.208.xx.yy > s=session > c=IN IP4 190.208.xx.yy > t=0 0 > m=audio 10236 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() > Audio Codec Compare [G729:18:8000:0]/[PCMU:0:8000:20] > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 sofia_glue_negotiate_sdp() > Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 sofia_handle_sip_i_state() Hangup > sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904 at 190.208.xx.yy [KILL] > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > send 886 bytes to udp/[190.47.91.83]:60245 at 16:19:20.596633: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.124:2051 > ;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > From: "102" ;tag=bmke36jc1v > To: "102" ;tag=rZp0XXrK9NHFD > Call-ID: 3c26700b249f-sryanqz0td8u at snom360-00041323143F > CSeq: 27056 REGISTER > Contact: ;line=dcnm5x2k>;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > Date: Mon, 01 Jun 2009 16:19:20 GMT > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running > State Change CS_HANGUP > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > HANGUP > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/admin/42452904 at 190.208.xx.yy hanging up, cause: > INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to > INVITE with: 488 > send 634 bytes to udp/[190.208.xx.yy]:5060 at 16:19:20.603208: > ------------------------------------------------------------------------ > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > From: "452904" ;tag=as4e2616ae > To: ;tag=S8FSZr9p6y71r > Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/admin/42452904 at 190.208.xx.yyStandard HANGUP, cause: INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > HANGUP going to sleep > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > Change CS_HANGUP -> CS_REPORTING > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) Running > State Change CS_REPORTING > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) > State REPORTING > recv 408 bytes from udp/[190.208.xx.yy]:5060 at 16:19:20.620338: > ------------------------------------------------------------------------ > ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 > Via: SIP/2.0/UDP 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > From: "452904" ;tag=as4e2616ae > To: ;tag=S8FSZr9p6y71r > Contact: > Call-ID: 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/admin/42452904 at 190.208.xx.yyStandard REPORTING, cause: INCOMPATIBLE_DESTINATION > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/42452904 at 190.208.xx.yy) > State REPORTING going to sleep > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/admin/42452904 at 190.208.xx.yy) State > Change CS_REPORTING -> CS_DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) > Locked, Waiting on external entities > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/admin/42452904 at 190.208.xx.yy) > Ended > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) > State DESTROY > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/admin/42452904 at 190.208.xx.yy SOFIA DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/admin/42452904 at 190.208.xx.yyStandard DESTROY > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/42452904 at 190.208.xx.yy) > State DESTROY going to sleep > > My vars.xml : > > data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > > > > I hope your comments for know where is the config problem > > Fernando. > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/41ebc073/attachment-0002.html From msc at freeswitch.org Mon Jun 1 10:16:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 10:16:26 -0700 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> Message-ID: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It's always august =D > it's ok. it counts as work! > If you need a letter to your boss we can do that on ClueCon letterhead. :) -MC > > > > On Mon, Jun 1, 2009 at 11:54 AM, bakko wrote: > >> Maybe next year. >> >> On August i have to work :(. >> >> If you organize ClueCon 2010 on setpember I wil go. >> >> :) >> >> Good luck. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3c59f9a9/attachment-0002.html From intralanman at freeswitch.org Mon Jun 1 10:19:53 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 01 Jun 2009 13:19:53 -0400 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl> <191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> Message-ID: <4A240DB9.3080208@freeswitch.org> Michael Collins wrote: > > > On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale > > wrote: > > It's always august =D > it's ok. it counts as work! > > > If you need a letter to your boss we can do that on ClueCon letterhead. :) > -MC > my boss paid me for the week last year when i went... any self-respecting employer can't see anything wrong with "Continuing Educational Opportunities" -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/3115ff57/attachment-0002.html From asannucci at gmail.com Mon Jun 1 10:29:09 2009 From: asannucci at gmail.com (bakko) Date: Mon, 1 Jun 2009 19:29:09 +0200 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl><191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> Message-ID: <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> I need 5 letters: - one for my boss - one for the boss of my boss - one for my wife - one for my bank (asking more credit) - one for me (like a post-it for don't forget the appointment) If you can do all this, I'will go :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/01d78450/attachment-0002.html From intralanman at freeswitch.org Mon Jun 1 10:33:11 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 01 Jun 2009 13:33:11 -0400 Subject: [Freeswitch-users] Problem about play wav file in conference In-Reply-To: <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> References: <200906010938513832282@163.com> <4A238F8E.8060400@gcdf.pl><191c3a030906011011v47a9b33x28d86dfd1e38ad49@mail.gmail.com> <87f2f3b90906011016q40358ed8t84533cc4b9fa7b9c@mail.gmail.com> <398ECB92EE8A40158C1A0D2AF7D4EBC6@voztovoice> Message-ID: <4A2410D7.8020309@freeswitch.org> > - one for my wife bring her too > - one for my bank (asking more credit) we can write it... but i won't guarantee that they give you what we ask :-P -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/6d012261/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Jun 1 12:33:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 20:33:37 +0100 Subject: [Freeswitch-users] Make current fails Message-ID: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Regards, making all mod_sndfile cd . && /bin/sh /usr/src/trunk/libs/libsndfile/missing --run automake-1.9 --gnu configure.ac: required file `Cfg/install-sh' not found configure.ac: required file `Cfg/missing' not found examples/Makefile.am: required file `Cfg/depcomp' not found programs/Makefile.am: required file `Cfg/compile' not found configure.ac:12: required file `Cfg/config.guess' not found configure.ac:12: required file `Cfg/config.sub' not found configure.ac:49: required file `Cfg/ltmain.sh' not found make[6]: *** [Makefile.in] Error 1 make[5]: *** [../../../../libs/libsndfile/src/libsndfile.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_sndfile-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/836db3e9/attachment-0002.html From brian at freeswitch.org Mon Jun 1 12:36:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 14:36:21 -0500 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: > Hi Guys, > > Been running ?make current? and appropriate intervals over the last > few months and all?s been well until today > > Now I get the following, obviously mod_sndfile isn?t happy, but I?m > not sure what to do to fix it > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/92d8d4b6/attachment-0002.html From e at musinghalfwit.org Mon Jun 1 13:12:06 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Mon, 1 Jun 2009 15:12:06 -0500 Subject: [Freeswitch-users] 300 Multiple Choices Message-ID: <20090601201206.GA6838@pointone.com> Hey all, Was curious if there is any 300 Multiple Choices support in freeswitch. It seems from the sofia logging that the sofia library just kills the call when it receives a 300 and freeswitch the same. Is this the case or am I missing something? I googled around and searched the list but didn't see anything definitive. So I just wanted to ensure I hadn't overlooked anything. -Eric From nik.middleton at noblesolutions.co.uk Mon Jun 1 13:52:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 21:52:14 +0100 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Thanks for that ./ bootstrap.sh ./configure Did the trick Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 20:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/84fcfa20/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Jun 1 14:18:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 22:18:28 +0100 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: Spoke too soon. Clean compile and install, but now FS hangs for about 5 mins on startup Error [unterminated ${var}] in file /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm l line 12 Error including /usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide character) The first error is a typo in the sample, but the second error, I don't have that DIR at all. I presume that this dir has been added, but how to I create these without overwriting my working configs? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 01 June 2009 21:52 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Thanks for that ./ bootstrap.sh ./configure Did the trick Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 20:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails Reboot strap. /b On Jun 1, 2009, at 2:33 PM, Nik Middleton wrote: Hi Guys, Been running 'make current' and appropriate intervals over the last few months and all's been well until today Now I get the following, obviously mod_sndfile isn't happy, but I'm not sure what to do to fix it Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/aabf16b3/attachment-0002.html From brian at freeswitch.org Mon Jun 1 14:29:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 16:29:15 -0500 Subject: [Freeswitch-users] Make current fails In-Reply-To: References: Message-ID: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> I can tell you how to fix it but it'll cost ya! :P /b > Spoke too soon. > > Clean compile and install, but now FS hangs for about 5 mins on > startup > > Error [unterminated ${var}] in file /usr/local/freeswitch/conf/ > autoload_configs/../jingle_profiles/client.xml line 12 > Error including /usr/local/freeswitch/conf/autoload_configs/../ > mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide > character) > > The first error is a typo in the sample, but the second error, I > don?t have that DIR at all. I presume that this dir has been added, > but how to I create these without overwriting my working configs? > > > Regards > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/f12eef55/attachment-0002.html From fvillarroel at yahoo.com Mon Jun 1 15:20:27 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 15:20:27 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <897953.58731.qm@web34301.mail.mud.yahoo.com> Hello the dial plan: This i setup from Wikipbx. --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 2:15 PM > What does your dialplan look like? Just > curious where/how you set proxy-media mode. > -MC > > On Mon, Jun 1, 2009 at 9:54 AM, > FERNANDO VILLARROEL > wrote: > > > > Dear all. > > > > I have problem with g729 passthru mode. > > > > I received traffic from a Asterisk on my FS and forward to > other Asterisk, when i use codec ulaw this works very well. > > > > But when i try use G729 i received the following messages > and SIP Trace: > > > > 2009-06-01 12:19:20 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel > sofia/admin/42452904 at 190.208.xx.yy > [f65514e0-4ec7-11de-9b78-150e2985561f] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3037 > sofia_handle_sip_i_state() Channel > sofia/admin/42452904 at 190.208.xx.yy entering state > [received][100] > > 2009-06-01 12:19:20 [DEBUG] sofia.c:3044 > sofia_handle_sip_i_state() Remote SDP: > > v=0 > > o=root 25643 25643 IN IP4 190.208.xx.yy > > s=session > > c=IN IP4 190.208.xx.yy > > t=0 0 > > m=audio 10236 RTP/AVP 18 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [G729:18:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2915 > sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 > > 2009-06-01 12:19:20 [DEBUG] sofia_glue.c:2955 > sofia_glue_negotiate_sdp() Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > > 2009-06-01 12:19:20 [NOTICE] sofia.c:3246 > sofia_handle_sip_i_state() Hangup > sofia/admin/42452904 at 190.208.xx.yy [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > 2009-06-01 12:19:20 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/42452904 at 190.208.xx.yy [KILL] > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > > send 886 bytes to udp/[190.47.91.83]:60245 at > 16:19:20.596633: > > ? > ------------------------------------------------------------------------ > > ? SIP/2.0 200 OK > > ? Via: SIP/2.0/UDP > 192.168.1.124:2051;branch=z9hG4bK-cod9x8o56t2q;rport=60245;received=190.47.91.83 > > ? From: "102" > ;tag=bmke36jc1v > > ? To: "102" > ;tag=rZp0XXrK9NHFD > > ? Call-ID: > 3c26700b249f-sryanqz0td8u at snom360-00041323143F > > ? CSeq: 27056 REGISTER > > ? Contact: > ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO";expires=30 > > > ? Date: Mon, 01 Jun 2009 16:19:20 GMT > > ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > ? Supported: timer, precondition, path, replaces > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) Running State Change > CS_HANGUP > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:323 > sofia_on_hangup() Channel sofia/admin/42452904 at 190.208.xx.yy > hanging up, cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:399 > sofia_on_hangup() Responding to INVITE with: 488 > > send 634 bytes to udp/[190.208.xx.yy]:5060 at > 16:19:20.603208: > > ? > ------------------------------------------------------------------------ > > ? SIP/2.0 488 Not Acceptable Here > > ? Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport=5060 > > ? From: "452904" > ;tag=as4e2616ae > > ? To: > ;tag=S8FSZr9p6y71r > > ? Call-ID: > 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > > ? CSeq: 102 INVITE > > ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13431 > > ? Accept: application/sdp > > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, > MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > > ? Supported: timer, precondition, path, replaces > > ? Allow-Events: talk, refer > > ? Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() > sofia/admin/42452904 at 190.208.xx.yy Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State HANGUP going to > sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State Change CS_HANGUP > -> CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/42452904 at 190.208.xx.yy [BREAK] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) Running State Change > CS_REPORTING > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING > > recv 408 bytes from udp/[190.208.xx.yy]:5060 at > 16:19:20.620338: > > ? > ------------------------------------------------------------------------ > > ? ACK sip:56968482060 at 200.111.XXX.XX SIP/2.0 > > ? Via: SIP/2.0/UDP > 190.208.xx.yy:5060;branch=z9hG4bK25dd52ee;rport > > ? From: "452904" > ;tag=as4e2616ae > > ? To: > ;tag=S8FSZr9p6y71r > > ? Contact: > > ? Call-ID: > 117330d21f3828470f39a95f538be036 at 190.208.xx.yy > > ? CSeq: 102 ACK > > ? User-Agent: Asterisk PBX > > ? Max-Forwards: 70 > > ? Content-Length: 0 > > > > ? > ------------------------------------------------------------------------ > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() > sofia/admin/42452904 at 190.208.xx.yy Standard REPORTING, > cause: INCOMPATIBLE_DESTINATION > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/42452904 at 190.208.xx.yy) State REPORTING going > to sleep > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() > (sofia/admin/42452904 at 190.208.xx.yy) State Change > CS_REPORTING -> CS_DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 7 > (sofia/admin/42452904 at 190.208.xx.yy) Locked, Waiting on > external entities > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 > (sofia/admin/42452904 at 190.208.xx.yy) Ended > > 2009-06-01 12:19:20 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/admin/42452904 at 190.208.xx.yy [CS_DESTROY] > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY > > 2009-06-01 12:19:20 [DEBUG] mod_sofia.c:240 > sofia_on_destroy() sofia/admin/42452904 at 190.208.xx.yy SOFIA > DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() > sofia/admin/42452904 at 190.208.xx.yy Standard DESTROY > > 2009-06-01 12:19:20 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/42452904 at 190.208.xx.yy) State DESTROY going to > sleep > > > > My vars.xml : > > > > ? data="global_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > ? data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,G729"/> > > ? data="xmpp_client_profile=xmppc"/> > > ? data="xmpp_server_profile=xmpps"/> > > > > > > I hope your comments for know where is the config problem > > > > Fernando. > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Mon Jun 1 15:24:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 1 Jun 2009 23:24:52 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> Message-ID: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 June 2009 22:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails I can tell you how to fix it but it'll cost ya! :P /b Spoke too soon. Clean compile and install, but now FS hangs for about 5 mins on startup Error [unterminated ${var}] in file /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xm l line 12 Error including /usr/local/freeswitch/conf/autoload_configs/../mrcp_profiles/*.xml (Invalid or incomplete multibyte or wide character) The first error is a typo in the sample, but the second error, I don't have that DIR at all. I presume that this dir has been added, but how to I create these without overwriting my working configs? Regards Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/1434dd7c/attachment-0002.html From brian at freeswitch.org Mon Jun 1 15:33:01 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 1 Jun 2009 17:33:01 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> Message-ID: <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: > Well I can only assume build 13537 is brain dead. Surely I > shouldn?t have to edit a whole bunch of configs to get it working. > FS now takes 3 minutes to start, with no indication as to what it?s > looking for in the logs. That said, to date ?make current? has > always worked well for me. Guess I was bound to hit a bad one > eventually. > > Still, it?s very frustrating. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/4790e407/attachment-0002.html From msc at freeswitch.org Mon Jun 1 15:41:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 1 Jun 2009 15:41:03 -0700 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <897953.58731.qm@web34301.mail.mud.yahoo.com> References: <897953.58731.qm@web34301.mail.mud.yahoo.com> Message-ID: <87f2f3b90906011541x6b27db62ra9231f2b2d9e92f9@mail.gmail.com> On Mon, Jun 1, 2009 at 3:20 PM, FERNANDO VILLARROEL wrote: > > Hello the dial plan: > > > > This i setup from Wikipbx. What about this in the dialplan? Or alternatively this in the SIP profile? I just want to make sure you're actually telling FS to use proxy media. If I may make a suggestion: use pastebin.freeswitch.org and pastebin the entire extension in the dialplan as well as a complete debug log of the call from the FS CLI. Please see this page for some handy tips on gathering information for troubleshooting: http://wiki.freeswitch.org/wiki/Reporting_Bugs -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/fc806580/attachment-0002.html From fvillarroel at yahoo.com Mon Jun 1 16:09:19 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 16:09:19 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <738513.5183.qm@web34308.mail.mud.yahoo.com> Hello i was try with: This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL > wrote: > > > > Hello the dial plan: > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fvillarroel at yahoo.com Mon Jun 1 16:10:06 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 1 Jun 2009 16:10:06 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <592820.20616.qm@web34305.mail.mud.yahoo.com> Hello i was try with: This is the log on FS_CLI: http://pastebin.freeswitch.org/9204 Fernando --- On Mon, 6/1/09, Michael Collins wrote: > From: Michael Collins > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 7:41 PM > > > On Mon, Jun 1, 2009 at 3:20 PM, > FERNANDO VILLARROEL > wrote: > > > > Hello the dial plan: > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > This i setup from Wikipbx. > What about this in the dialplan? > data="proxy_media=true"/> > Or alternatively this in the SIP profile? > > value="true"/> > > I just want to make sure you're actually telling FS to > use proxy media. If I may make a suggestion: use pastebin.freeswitch.org > and pastebin the entire extension in the dialplan as well as > a complete debug log of the call from the FS CLI. Please see > this page for some handy tips on gathering information for > troubleshooting: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brad.tuan at gmail.com Mon Jun 1 20:02:42 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 11:02:42 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: This question can be separated into two part: 1.Pass a call to another FS 2.Receive a call from another FS Somebody can tell me how to do these?? Please............. -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/70a2f185/attachment-0002.html From jcromes at gmail.com Mon Jun 1 20:22:05 2009 From: jcromes at gmail.com (j3flight) Date: Mon, 1 Jun 2009 20:22:05 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> <4A21E87E.90709@gmail.com> <191c3a030906010608x3a11341cie9420912f6fc4645@mail.gmail.com> Message-ID: <23825864.post@talk.nabble.com> You rock! Thanks for doing that, makes things work nicely. Although, I was in the process of putting together a php event socket thingy to handle conference state changes. I didn't get very far yet, but it was a good exercise! I do appreciate your work - break granted. Jason -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23825864.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Jun 1 20:23:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 1 Jun 2009 22:23:03 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <191c3a030906012023l7f7dd608k33648dac8f52119e@mail.gmail.com> Try using a telephone? On Mon, Jun 1, 2009 at 10:02 PM, Brad Tuan wrote: > This question can be separated into two part: > 1.Pass a call to another FS > > 2.Receive a call from another FS > Somebody can tell me how to do these?? > > Please............. > > -Brad > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/399edc79/attachment-0002.html From jason at jasonjgw.net Mon Jun 1 20:27:04 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 13:27:04 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602032704.GA5858@jdc.jasonjgw.net> Brad Tuan wrote: > This question can be separated into two part: > 1.Pass a call to another FS uuid_deflect or uuid_transfer, depending on whether the call has been answered by the first FS instance or not. See the wiki. > > 2.Receive a call from another FS Provide a dial plan entry in the second FS that handles the call appropriately after the deflect or transfer. You haven't explained what you're trying to do - a general question warrants a general answer. I am assuming the call is arriving at one FS system, and (before or after answering it), you want to move it to another FS system. That's the question I've answered above. The wiki documents the syntax. From b_ball_henry at hotmail.com Mon Jun 1 20:46:54 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Mon, 1 Jun 2009 20:46:54 -0700 Subject: [Freeswitch-users] Polycom Phone issue Message-ID: <59ad9ca10906012046q48db6b86v69889aafff8ef0ab@mail.gmail.com> I have setup 2 Freeswitch test server, 2 IP430 polycom phones with 2 lines each registered to different server. When I dial extension from the first line (first FS server), if the called party hit reject on the phone. The called won't be hanged up will keep ringing until timed out. When dialed from second line (2nd FS server), the other phone would ring on the screen as of the first line is ringing and besides the caller-id and caller-id-name, it will also show the sip: address of the caller. Now it has similar sympton to the above senario, but more over, even if leg B picks up the call then hang up, Leg A (caller's end) will not hang up for as long as you don't hang it up. Do any of you people with polycom phone have problem like this ? or know what could be the cause? my FS1 server is version 12242M , FS2 server version is 13523M *Here is my dialplan example of the extension that I called:* * And here is the setting for register 2 server on polycom phone:* reg.1.displayName="2025" reg.1.address="2025" reg.1.label="2025" reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="" reg.1.auth.userId="2025" reg.1.auth.password="somepassword" reg.1.server.1.address="10.48.5.83" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.expires.overlap="" reg.1.server.1.register="1" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.outboundProxy.address="" reg.1.outboundProxy.port="" reg.1.outboundProxy.transport="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.proxyRequire="" reg.1.ringType="12" reg.1.lineKeys="1" reg.1.callsPerLineKey="" reg.2.displayName="2025" reg.2.address="2025" reg.2.label="2025" reg.2.type="private" reg.2.lcs="" reg.2.thirdPartyName="" reg.2.auth.userId="2025" reg.2.auth.password="somepassword" reg.2.server.1.address="10.55.7.36" reg.2.server.1.port="5060" reg.2.server.1.transport="DNSnaptr" reg.2.server.2.transport="DNSnaptr" reg.2.server.1.expires="" reg.2.server.1.expires.overlap="" reg.2.server.1.register="" reg.2.server.1.retryTimeOut="" reg.2.server.1.retryMaxCount="" reg.2.server.1.expires.lineSeize="" reg.2.outboundProxy.address="" reg.2.outboundProxy.port="" reg.2.outboundProxy.transport="" reg.2.acd-login-logout="0" reg.2.acd-agent-available="0" -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090601/47d30482/attachment-0002.html From brad.tuan at gmail.com Mon Jun 1 20:47:54 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 11:47:54 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: >>* This question can be separated into two part: *>>* 1.Pass a call to another FS * >uuid_deflect or uuid_transfer, depending on whether the call has been answered >by the first FS instance or not. See the wiki. >>* *>>* 2.Receive a call from another FS * >Provide a dial plan entry in the second FS that handles the call appropriately >after the deflect or transfer. >You haven't explained what you're trying to do - a general question warrants a >general answer. I am assuming the call is arriving at one FS system, and >(before or after answering it), you want to move it to another FS system. >That's the question I've answered above. The wiki documents the syntax. SorrySorry,let me explain my question. When User1( User of FS1 ) call User2( User of FS2 ) , FS1 will pass the call to FS2 before answering, and then User1 can talk with User2. -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/87cf3686/attachment-0002.html From jason at jasonjgw.net Mon Jun 1 21:04:38 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 14:04:38 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602040438.GA10771@jdc.jasonjgw.net> Brad Tuan wrote: > When User1( User of FS1 ) call User2( User of FS2 ) , > > FS1 will pass the call to FS2 before answering, You just need to write a dial plan extension that matches the call on FS1 and bridges it to FS2. For example: Dialing any extension with the prefix 014 will call that extension (with the prefix removed) on fs2.example.org at port 5080. If FS2 has the default configuration installed, the call will land in the public context of fs2, where you can transfer it to the default context or take other actions depending on the extension called. From brad.tuan at gmail.com Mon Jun 1 22:35:51 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 13:35:51 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: I have tried But the console moniter return : [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187:5080 at external [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED I am sure that 97710001 is a online user on "192.168.141.187", What's wrong?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/412e6cfb/attachment-0002.html From brian at freeswitch.org Mon Jun 1 22:44:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 00:44:56 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <4C970DD9-01C1-4781-9E9B-F67C8401F769@freeswitch.org> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings If you're trying to call a registered user that is NOT the way to do it. sofia/profile/user%domain /b On Jun 2, 2009, at 12:35 AM, Brad Tuan wrote: > I have tried > > > > > > > > But the console moniter return : > [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001- > >01497710001 in context default > [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate > registered user 97710001 at 192.168.141.187:5080 at external > [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A > [CS_NEW] > [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot > create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. > Cause: USER_NOT_REGISTERED > > I am sure that 97710001 is a online user on "192.168.141.187", > What's wrong?? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/7d7357e6/attachment-0002.html From jason at jasonjgw.net Mon Jun 1 22:58:29 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 15:58:29 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602055829.GA29143@jdc.jasonjgw.net> Brad Tuan wrote: > I have tried > > > > > > > > But the console moniter return : > [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 > in context default > [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot locate registered > user 97710001 at 192.168.141.187:5080 at external > [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A > [CS_NEW] > [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create > outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: > USER_NOT_REGISTERED > > I am sure that 97710001 is a online user on "192.168.141.187", What's > wrong?? It will work if you use a domain name for the host rather than an IP address. From jason at jasonjgw.net Mon Jun 1 23:05:08 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 16:05:08 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602060508.GA30762@jdc.jasonjgw.net> Also, if the other FS box is behind the same NAT you're on, you should be using the internal profile: sofia/internal/$1 at 192.168.xxx.xxx or whatever. From plite2012 at gmail.com Mon Jun 1 23:12:38 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 2 Jun 2009 01:12:38 -0500 Subject: [Freeswitch-users] How to specify the path to fax file on Windows? Message-ID: All the examples related to faxing use the Unix/Linux path, such as originate sofia/external/100 at 10.10.10.10 &txfax(/path_to_fax_file) I have tried "C:/tmp/fax/txfax.tiff" or "C:\MyJob\fax\txfax.tiff" or "C:\\MyJob\\fax\\txfax.tiff" without any luck. I got an error like [ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file, or the app crashed. Any help is greatly appreciated! From brad.tuan at gmail.com Mon Jun 1 23:18:22 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 14:18:22 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: Like this?? *> *>* *>* * >* *>* * I have tried,but Fs still return: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->01497710001 in context default [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001 at 192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/50e544d5/attachment-0002.html From brad.tuan at gmail.com Mon Jun 1 23:37:13 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 14:37:13 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? Message-ID: I have tried But FS still return the same message........ [WARNING] mod_sofia.c:2534 sofia_outgoing_channel() Cannot locate registered user 97710001 at 192.168.141.187 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Channel N/A [CS_NEW] [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] [INFO] mod_dptools.c:1998 audio_bridge_function() OriginateFailed. Cause: USER_NOT_REGISTERED [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1001 at 192.168.141.182 [CS_EXECUTE] [USER_NOT_REGISTERED] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/231aef43/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 01:04:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 16:04:49 +0800 Subject: [Freeswitch-users] How to reload xml without using console command line?? Message-ID: As title How to reload xml without using console command line?? -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/db106a0b/attachment-0002.html From jason at jasonjgw.net Tue Jun 2 01:22:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 18:22:56 +1000 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: <20090602082256.GA15770@jdc.jasonjgw.net> Brad Tuan wrote: > As title Write a script that connects to the event socket and issues an api reloadxml command. From jason at jasonjgw.net Tue Jun 2 01:23:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 2 Jun 2009 18:23:56 +1000 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: <20090602082356.GB15770@jdc.jasonjgw.net> Brad Tuan wrote: > I have tried > > > > Change the % to an @ in the above. From brad.tuan at gmail.com Tue Jun 2 02:00:29 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 17:00:29 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: <20090602082356.GB15770@jdc.jasonjgw.net> References: <20090602082356.GB15770@jdc.jasonjgw.net> Message-ID: the same message........ 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 100 1->97710001 in context default 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot l ocate registered user 97710001 at 192.168.141.187 at internal 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED 2009/6/2 Jason White > Brad Tuan wrote: > > I have tried > > > > > > > > data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/e6737222/attachment-0002.html From krice at suspicious.org Tue Jun 2 02:12:47 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 02 Jun 2009 04:12:47 -0500 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: Message-ID: Dumb question... Is 187 the local fs machine? You should have the IP address of the remote FS machine From: Brad Tuan Reply-To: Date: Tue, 2 Jun 2009 17:00:29 +0800 To: Subject: Re: [Freeswitch-users] How to pass a call from one FS to another FS ?? the same message........ ? 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 100 1->97710001 in context default 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() Cannot l ocate registered user 97710001 at 192.168.141.187 at internal 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Can not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed.? Cause: USER_NOT_REGISTERED ? 2009/6/2 Jason White > Brad Tuan wrote: >> > I have tried >> > >> > >> > ? ? >> > ? ? ?> data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/abfdfc3f/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 02:25:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 17:25:04 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 These two FS are in the same LAN. I just try to pass one sip call from one FS to another. If it works, next is FS1( PublicIP ) to FS2( PublicIP ). 2009/6/2 Ken Rice > Dumb question... Is 187 the local fs machine? You should have the IP > address of the remote FS machine > > > ------------------------------ > *From: *Brad Tuan > *Reply-To: * > *Date: *Tue, 2 Jun 2009 17:00:29 +0800 > *To: * > *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to > another FS ?? > > > the same message........ > > 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 100 > 1->97710001 in context default > 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() > Cannot l > ocate registered user 97710001 at 192.168.141.187 at internal > 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() > Close Cha > nnel N/A [CS_NEW] > 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Can > not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate > Failed. Cause: USER_NOT_REGISTERED > > 2009/6/2 Jason White > > Brad Tuan wrote: > > I have tried > > > > > > > > data="sofia/internal/$1%192.168.141.187"/> > > Change the % to an @ in the above. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/3a2eee2a/attachment-0002.html From freeswitch at davidnicol.otherinbox.com Tue Jun 2 03:43:52 2009 From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com) Date: Tue, 2 Jun 2009 06:43:52 -0400 Subject: [Freeswitch-users] Make current fails (build 13537) Message-ID: <200906021043.n52AhqKE005179@box7.911domain.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/af746b3f/attachment-0002.html From shaheryarkh at googlemail.com Tue Jun 2 03:40:02 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 2 Jun 2009 16:40:02 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up Message-ID: Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/ee029b30/attachment-0002.html From yivzhenko at mksat.net Tue Jun 2 04:04:06 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 2 Jun 2009 14:04:06 +0300 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring Message-ID: <200906021404.07496.yivzhenko@mksat.net> Hello all. I have test the lcr overriding the Caller ID functionality. It return dialstring, that contains 'effective_caller_id_number' variable. But that variable has no effect. I try test configuration There is no result. (caller id number not changed) But If I uncomment the set line, then the caller_id_number changes. I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and his status - fixed. ....may be i not consider something? I use svn trunk 13544. Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/780c315c/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 04:08:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 2 Jun 2009 12:08:52 +0100 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: As I understand it, a new 'feature' was added over the weekend to resolve NAT. If you're firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/932019d8/attachment-0002.html From dave at 3c.co.uk Tue Jun 2 04:23:07 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 2 Jun 2009 12:23:07 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: At the risk of evisceration (but with the intention of helping avoid future brain dead build vs. idiot admin debates), I'd suggest that, when significant new bits are added to the switch core, they should default to being off and require a configuration option to turn them on. Such config options can be added to the default config; that way new installs will have the new functionality enabled by default, but those upgrading from an older install will need to enable them manually, reducing the risk of stuff breaking. --Dave ----- Original Message ----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Monday, June 01, 2009 11:33 PM Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn?t have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it?s looking for in the logs. That said, to date ?make current? has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it?s very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/c7159f4d/attachment-0002.html From dujinfang at gmail.com Tue Jun 2 04:27:53 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 19:27:53 +0800 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > As I understand it, a new ?feature? was added over the weekend to > resolve NAT. If you?re firewall is not allowing ICMP then FS waits > until it times out. At this time there is no option to disable it. > > Regards > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Muhammad Shahzad > Sent: 02 June 2009 11:40 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Freeswitch taking too long to start up > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I > am using 32bit CentOS 5.3, "make current" command completes > successfully without any errors but when i start freeswitch it take > considerable time (roughly 90 - 120 seconds) to start up. During > this time no message is display on console. Once successfully > started, it works fine. However, this initial delay is really > annoying. Is there anyway to reduce/remove this delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/468e97b7/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 04:28:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 19:28:04 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call Message-ID: Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"?? EVENT DUMP: Channel-State: [CS_ROUTING] Channel-State-Number: [2] Channel-Name: [sofia/internal/sip:97730001 at 210.68.184.192:62101 ;rinstance=16b8076934af7da9] Unique-ID: [342618e3-84cd-494b-b745-760b60639924] Call-Direction: [outbound] Answer-State: [ringing] Caller-Username: [97719006] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Extension 97730002] Caller-Caller-ID-Number: [97730002] Caller-Network-Addr: [163.28.32.51] Caller-Destination-Number: [sip:97730001 at 210.68.184.192:62101 ;rinstance=16b80769 34af7da9] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/879ab698/attachment-0002.html From codecomplete at free.fr Tue Jun 2 04:40:55 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 2 Jun 2009 04:40:55 -0700 (PDT) Subject: [Freeswitch-users] Can Freeswitch + LAMP run on 128MB RAM? In-Reply-To: <4A23D15E.3040908@coppice.org> References: <23807353.post@talk.nabble.com> <4A23D15E.3040908@coppice.org> Message-ID: <23830858.post@talk.nabble.com> Steve Underwood wrote: > Nobody has yet adapted Freeswitch for the Blackfin, and they probably > won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux, > which is a cut down Linux for machines of this type. It is quite > troublesome to get memory management to behave sanely on a machine > without an MMU. The Asterisk adaptions for the Blackfin have problems > with this too, but if you don't let the memory become too fragmented they > work OK. Thanks much for the explanation. I don't need the fax module. Hopefully, other features will work fine on this unit. I haven't found other hardware that is as compact and affordable as the Atcom while providing an embedded FXO port. -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p23830858.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From juanbackson at gmail.com Tue Jun 2 05:01:05 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 2 Jun 2009 20:01:05 +0800 Subject: [Freeswitch-users] some fifo questions Message-ID: <27c25bc40906020501xd51fea2x49949b8855306751@mail.gmail.com> Hi, I read the fifo section of the wiki and what is not clear are: What is the meaning of fifo_orbit_announce? What is the meaning of fifo_override_announce? Is it possible to create a scenario where the caller can hear "Agent #123 is going to attend to your call"? Any help will be greatly appreciated. Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/02c2ec6c/attachment-0002.html From jim at evolutiontel.net Tue Jun 2 05:25:38 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 2 Jun 2009 22:25:38 +1000 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> <191c3a030905280607q7dd0b5b7xc9e3f02b9f6c8824@mail.gmail.com> <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> Message-ID: Hey Gents, What is the Jira for this issue? Dale, Did you get any SIP traces. I am interested to have a look. You can use NGREP if your system is Linux. Regards, Jim On Thu, May 28, 2009 at 11:08 PM, Anthony Minessale wrote: > btw, > > ?3 and 4 are not useful without 1 > we only debug issues with svn trunk > > > On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale > wrote: >> >> Also you should be putting these details in a jira report. >> http://jira.freeswitch.org >> >> open an issue report and attach all relevant logs, do not attach tarballs >> or gzipped files and make sure text files have a .txt extension. >> >> >> On Wed, May 27, 2009 at 6:58 PM, Dale Trub wrote: >>> >>> Anthony, >>> >>> Thank you for your suggestions!? We are working on 1), but need to >>> re-integrate code we've changed, and do regression testing. That's in >>> progress, and we expect to be able to upgrade by the end of next week. >>> >>> We did manage to do 3) and 4), and we now have SIP logs (attached). Are >>> you able to see anything that's out of the ordinary that we should be paying >>> attention to? >>> >>> Best, >>> Dale >>> >>> On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale >>> wrote: >>>> >>>> 1) update to lastest trunk (you are at least 1000 revisions behind) >>>> 2) disable the presence debug in sofia.conf >>>> 3) enable sip trace instead "sofia profile internal siptrace on" >>>> 4) reproduce your problem. >>>> >>>> Make sure you include more of the log from before the hangup happened. >>>> The one you posted here is missing some of the info from the few seconds >>>> prior but with the incomplete >>>> info it looks like the other side sent a BYE ending the call. >>>> >>>> >>>> On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: >>>>> >>>>> Thanks Brian! ?To answer your questions: >>>>> Freeswitch svn revision: 12148 >>>>> Centos rev: 2.6.18-92.el5 >>>>> And apologies, actually I guess we're using g711 not 729. >>>>> Jason: ?I agree it would seem to be on the switch/telco side. ?And, the >>>>> telco says many other people are in the same set-up as us and don't have any >>>>> issues, so they're insisting it's on our end. >>>>> On Thu, May 21, 2009 at 7:28 PM, Brian West >>>>> wrote: >>>>>> >>>>>> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >>>>>> >>>>>> We're running FreeSwitch as part of a teleconferencing service, inside >>>>>> a telcom?(so no >>>>>> internet latency/NAT issues)?and using g.729 >>>>>> >>>>>> So you're using g729 with conferences? >>>>>> >>>>>> We are?receiving some complaints of dropped calls, >>>>>> including from landlines. ? This means they join the conference, and x >>>>>> minutes in they simply drop. >>>>>> I?know that cellphones tend to drop calls frequently, but landlines >>>>>> are pretty reliable, and we're hearing it a lot. ?From the FreeSwitch >>>>>> side of things, it just >>>>>> looks like those callers hung up (but then dialed back in just a >>>>>> moment later). >>>>>> I'm attaching two different snippets of the FS log files where these >>>>>> issues are occurring. >>>>>> >>>>>> Next time please call them .txt because you cause extra work to have >>>>>> to open them otherwise. >>>>>> >>>>>> Does anyone have any recommendations about how to troubleshoot this? >>>>>> Any known issues/patches in FS that could be biting us? >>>>>> >>>>>> Depends you failed to include some very valid info such as what >>>>>> version or svn rev you're running and what linux distro. >>>>>> >>>>>> Is there some SIP logging we can do to debug? >>>>>> >>>>>> Yes covered on the wiki. >>>>>> ?http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>>>> >>>>>> Are there any paid contractors avail who would have the expertise to >>>>>> look into this? >>>>>> >>>>>> email consulting at freeswitch.org >>>>>> >>>>>> Any help appreciated ... this is a major issue for us! >>>>>> Thanks much, >>>>>> -Dale >>>>>> >>>>>> Brian West >>>>>> brian at freeswitch.org >>>>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Tue Jun 2 05:38:30 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 2 Jun 2009 18:38:30 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: > Actually Brain mentioned that you can comment out switch_nat_init(); in > switch_core.c > On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > > As I understand it, a new ?feature? was added over the weekend to resolve > NAT. If you?re firewall is not allowing ICMP then FS waits until it times > out. At this time there is no option to disable it. > > Regards > > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Muhammad Shahzad > *Sent:* 02 June 2009 11:40 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch taking too long to start up > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I am > using 32bit CentOS 5.3, "make current" command completes successfully > without any errors but when i start freeswitch it take considerable time > (roughly 90 - 120 seconds) to start up. During this time no message is > display on console. Once successfully started, it works fine. However, this > initial delay is really annoying. Is there anyway to reduce/remove this > delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/3fccb446/attachment-0002.html From brian at freeswitch.org Tue Jun 2 05:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 07:52:10 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: Its coming soon! /b On Jun 2, 2009, at 6:23 AM, David Knell wrote: > At the risk of evisceration (but with the intention of helping avoid > future brain dead build vs. idiot admin debates), I'd suggest that, > when significant new bits are added to the switch core, they should > default to being off and require a configuration option to turn them > on. Such config options can be added to the default config; that > way new installs will have the new functionality enabled by default, > but those upgrading from an older install will need to enable them > manually, reducing the risk of stuff breaking. > > --Dave Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/fe248b97/attachment-0002.html From brian at freeswitch.org Tue Jun 2 05:52:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 07:52:46 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: Chances are that is what you set it to on the user. Verify the users settings in the directory. /b On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: > Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name > is "Extension 97730002"?? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/df2565bb/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 06:11:07 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:11:07 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: I don't have a 97719006 User in my FS. It was passed from another sip proxy. 2009/6/2 Brian West > Chances are that is what you set it to on the user. Verify the users > settings in the directory. > /b > > On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: > > Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is > "Extension 97730002"?? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/b8f033b0/attachment-0002.html From brian at freeswitch.org Tue Jun 2 06:26:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 08:26:16 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: Then it had to be passed in from the proxy. /b On Jun 2, 2009, at 8:11 AM, Brad Tuan wrote: > I don't have a 97719006 User in my FS. > > It was passed from another sip proxy. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/70456bce/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 06:28:58 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:28:58 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: When send 100 Trying: From: 97719006 ;tag=124388224932run00 But when send INVITE: From: "Extension 97730002" >;tag=U3QF8QUp1F3tQ What happened between sending Trying and sending INVITE ?? ------------------------------------------------------------------------ send 556 bytes to udp/[163.28.32.51]:5070 at 13:03:05.218750: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK5b64.83b3e3d3.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12439477847278735900 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124394778472787359 Record-Route: From: 97719006 ;tag=124388224932run00 To: Call-ID: i58YWNjMDU3ZWJhN2M1YzVlYjMzOTgxMjk4OWZiNTU0Yzc.00 CSeq: 7359 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/97719006 at 61.61.162.13 -b28d-fd1ecda44f18] 2009-06-02 21:03:05 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 97719006->97730001 in context default 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::C:SipG -06-02-21-03-05.wav 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:97730001 at 210.68.1 d6224d69efcf1 [e8ef86f8-e5aa-d246-8ffb-bf9e0f9dc160] send 1347 bytes to udp/[210.68.184.192]:62113 at 13:03:05.812500: ------------------------------------------------------------------------ INVITE sip:97730001 at 210.68.184.192:62113;rinstance=d0ed6224d69efcf1 SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKg01ymegmSyp5c Max-Forwards: 67 From: "Extension 97730002" >;tag=U3QF8QUp1F3tQ To: Call-ID: 8dcc44c8-ca18-122c-2780-39a48cb53b8d CSeq: 115855556 INVITE 2009/6/2 Brad Tuan > I don't have a 97719006 User in my FS. > > It was passed from another sip proxy. > > > 2009/6/2 Brian West > >> Chances are that is what you set it to on the user. Verify the users >> settings in the directory. >> /b >> >> On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote: >> >> Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is >> "Extension 97730002"?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/10c48ae7/attachment-0002.html From brian at freeswitch.org Tue Jun 2 06:33:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 08:33:05 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: I would update if I were you! :) Anyway something had to have changed it it won't magically do it. /b On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/31589174/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 06:38:45 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:38:45 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: So I need to new a User(97719006) in directory\default ?? 2009/6/2 Brian West > I would update if I were you! :) Anyway something had to have changed it > it won't magically do it. > /b > > On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/fc31d939/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 06:41:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 2 Jun 2009 21:41:04 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: How to update FreeSWITCH-mod_sofia/1.0.3-12163?? 2009/6/2 Brian West > I would update if I were you! :) Anyway something had to have changed it > it won't magically do it. > /b > > On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote: > > User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f4bcc499/attachment-0002.html From rex.alex345 at yahoo.com Tue Jun 2 07:14:13 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 2 Jun 2009 07:14:13 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS Message-ID: <1243952053200-3012286.post@n2.nabble.com> Hello, I have installed FS, and tested outbound successfully. Now I am just trying to do the inbound testing. I got the Inbound DID. Please suggest me what changes should I make and where? Thanks, Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012286.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/bdc5fce0/attachment-0002.html From dujinfang at gmail.com Tue Jun 2 07:27:00 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:27:00 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: Hi, I always got 0 messages when using web. Finally I added some debug information in the code and get this: 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3883 voicemail_api_function() port:[8080] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3884 voicemail_api_function() uri:[/domains/192.168.1.16/api/voicemail] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3885 voicemail_api_function() user:[] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3886 voicemail_api_function() domain:[192.168.1.16] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3887 voicemail_api_function() path_info:[web] It seems the freeswitch-user header not set by xml_rpc and user = switch_event_get_header(stream->param_event, "freeswitch-user"); cannot get the user. Any idea? Thanks. From dujinfang at gmail.com Tue Jun 2 07:30:27 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:30:27 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: <35216FF7-6120-4F44-84B5-25F1540A93C3@gmail.com> sorry forgot to mention I'm on FreeSWITCH Version 1.0.trunk (13524M) From d at unwire.it Tue Jun 2 07:35:59 2009 From: d at unwire.it (Darin Weeks) Date: Tue, 2 Jun 2009 07:35:59 -0700 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243952053200-3012286.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> Message-ID: <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> Have you setup an inbound gateway similar to this? http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing You also need to setup your dialplans for the inbound.... this page among others has more info: http://wiki.freeswitch.org/wiki/Quick_Start Finally, your FIREWALL can be the most critical to get right. In fact, you might want to start with this or revisit even if you think it is setup right. I was grasping around in the dark until I started using TCPDUMP to monitor what was happening with my connections. In the end, I realized that I needed to open certain ports on the SENDING side -- so, for example, calls coming FROM port 5060 to *any* port on my side I actually forward to port 5080 on my freeswitch server. At least I think that's what I ended up doing, but there are several different rules I setup as well. On Tue, Jun 2, 2009 at 7:14 AM, Rex_Alex wrote: > Hello, I have installed FS, and tested outbound successfully. Now I am just > trying to do the inbound testing. I got the Inbound DID. Please suggest me > what changes should I make and where? Thanks, Rex > ------------------------------ > View this message in context: Inbound using FS > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/4dd04e3a/attachment-0002.html From dujinfang at gmail.com Tue Jun 2 07:36:34 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 2 Jun 2009 22:36:34 +0800 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243952053200-3012286.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> Message-ID: <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> I would route the DID to the host and port 5080 if you are using the default config, and make an extension in dialplan/public.xml to catch the DID. Press F8 to see the debug information if not sure what DID string should be matched. On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote: > Hello, I have installed FS, and tested outbound successfully. Now I > am just trying to do the inbound testing. I got the Inbound DID. > Please suggest me what changes should I make and where? Thanks, Rex > View this message in context: Inbound using FS > Sent from the freeswitch-users mailing list archive at Nabble.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/dbddbbe7/attachment-0002.html From d at unwire.it Tue Jun 2 07:39:00 2009 From: d at unwire.it (Darin Weeks) Date: Tue, 2 Jun 2009 07:39:00 -0700 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <989132e70906020735l7ef37e9ake7d7955b44e154c8@mail.gmail.com> Message-ID: <989132e70906020739n2e7084dm2f3e57d1555f20e2@mail.gmail.com> UPDATE: I just looked at my firewall rules and looks like I scrapped all the logic I was attempting and now I just port forward ANYTHING coming from the IP of my provider gateway to my freeswitch box. Seems to be working fine. On Tue, Jun 2, 2009 at 7:35 AM, Darin Weeks wrote: > Have you setup an inbound gateway similar to this? > http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing > > You also need to setup your dialplans for the inbound.... this page among > others has more info: > http://wiki.freeswitch.org/wiki/Quick_Start > > Finally, your FIREWALL can be the most critical to get right. In fact, you > might want to start with this or revisit even if you think it is setup > right. I was grasping around in the dark until I started using TCPDUMP to > monitor what was happening with my connections. In the end, I realized that > I needed to open certain ports on the SENDING side -- so, for example, calls > coming FROM port 5060 to *any* port on my side I actually forward to port > 5080 on my freeswitch server. At least I think that's what I ended up > doing, but there are several different rules I setup as well. > > On Tue, Jun 2, 2009 at 7:14 AM, Rex_Alex wrote: > >> Hello, I have installed FS, and tested outbound successfully. Now I am >> just trying to do the inbound testing. I got the Inbound DID. Please suggest >> me what changes should I make and where? Thanks, Rex >> ------------------------------ >> View this message in context: Inbound using FS >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/6aa5e361/attachment-0002.html From mike at jerris.com Tue Jun 2 08:59:32 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Jun 2009 11:59:32 -0400 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: /usr/local/freeswitch/bin/fs_cli -x reloadxml On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote: > How to reload xml without using console command line?? From mike at jerris.com Tue Jun 2 09:04:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 2 Jun 2009 12:04:53 -0400 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <200906021404.07496.yivzhenko@mksat.net> References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: <07772B7F-6DCD-4D70-A3F9-AE861CEB6E29@jerris.com> can you try in the square brackets using http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number instead? I think effective will work if you set it but not in the square brackets. Mike On Jun 2, 2009, at 7:04 AM, Yuriy Ivzhenko wrote: > Hello all. > I have test the lcr overriding the Caller ID functionality. > It return dialstring, that contains 'effective_caller_id_number' > variable. > But that variable has no effect. > I try test configuration > > > > data="[effective_caller_id_number=9999]sofia/internal/sip:1001 at 192.168.2.43:5060 > "/> > > > > There is no result. (caller id number not changed) > But If I uncomment the set line, then the caller_id_number changes. > I found the similar bug (http://jira.freeswitch.org/browse/ > MODAPP-122) and his status - fixed. > ....may be i not consider something? > I use svn trunk 13544. > Yuriy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/7c3ddb15/attachment-0002.html From rex.alex345 at yahoo.com Tue Jun 2 09:11:50 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 2 Jun 2009 09:11:50 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> Message-ID: <1243959110713-3012928.post@n2.nabble.com> Hello, My public.xml configration is: My default.xml configration is: When I am trying to call 123456 from my mobile no. Not able to see any logging in FS console. Please assist where I am going wrong? Or do I require any extra modules to be installed? Thanks, Rex dujinfang wrote: > > I would route the DID to the host and port 5080 if you are using the > default config, and make an extension in dialplan/public.xml to catch > the DID. Press F8 to see the debug information if not sure what DID > string should be matched. > > > On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote: > >> Hello, I have installed FS, and tested outbound successfully. Now I >> am just trying to do the inbound testing. I got the Inbound DID. >> Please suggest me what changes should I make and where? Thanks, Rex >> View this message in context: Inbound using FS >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012928.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 2 09:15:52 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 11:15:52 -0500 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1243959110713-3012928.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> Message-ID: On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > > Hello, > > My public.xml configration is: > > > > > > $1 will not exist in this case because your regular expression doesn't capture anything. So replace $1 with your target number or use ^(123456)$ > > My default.xml configration is: > > > > > > > > Can you elaborate how you're registering with your provider? > > > When I am trying to call 123456 from my mobile no. Not able to see any > logging in FS console. Please assist where I am going wrong? Or do I > require > any extra modules to be installed? > > Thanks, > Rex Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/98113acd/attachment-0002.html From rdenert at tng.de Tue Jun 2 09:18:49 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 2 Jun 2009 18:18:49 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <2613803.271261243959087777.JavaMail.root@zimbra.tng.de> Message-ID: <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Greetz From dujinfang at gmail.com Tue Jun 2 09:30:14 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 00:30:14 +0800 Subject: [Freeswitch-users] always got 0 messages when retrieving voicemail from web Message-ID: I thought it is a problem, made a jira: http://jira.freeswitch.org/browse/XML-2 From rupa at rupa.com Tue Jun 2 09:40:12 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 2 Jun 2009 11:40:12 -0500 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <200906021404.07496.yivzhenko@mksat.net> References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: I've fixed mod_lcr now. It should have been setting origination_caller_id_number not effective_caller_id_number. On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > Hello all. > > I have test the lcr overriding the Caller ID functionality. > > It return dialstring, that contains 'effective_caller_id_number' variable. > > But that variable has no effect. > > I try test configuration > > > > > > > > data="[effective_caller_id_number=9999]sofia/internal/ > sip:1001 at 192.168.2.43:5060"/> > > > > > > > > There is no result. (caller id number not changed) > > But If I uncomment the set line, then the caller_id_number changes. > > I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and > his status - fixed. > > ....may be i not consider something? > > I use svn trunk 13544. > > Yuriy > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/14779423/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 2 09:50:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 11:50:13 -0500 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: References: <200906021404.07496.yivzhenko@mksat.net> Message-ID: <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> effective_* is *NOT EVER* valid in the dial string. they are settings of an existing session to control what caller id they pass. On Tue, Jun 2, 2009 at 11:40 AM, Rupa Schomaker wrote: > I've fixed mod_lcr now. It should have been setting > origination_caller_id_number not effective_caller_id_number. > > On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > >> Hello all. >> >> I have test the lcr overriding the Caller ID functionality. >> >> It return dialstring, that contains 'effective_caller_id_number' variable. >> >> But that variable has no effect. >> >> I try test configuration >> >> >> >> >> >> >> >> > data="[effective_caller_id_number=9999]sofia/internal/ >> sip:1001 at 192.168.2.43:5060"/> >> >> >> >> >> >> >> >> There is no result. (caller id number not changed) >> >> But If I uncomment the set line, then the caller_id_number changes. >> >> I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) >> and his status - fixed. >> >> ....may be i not consider something? >> >> I use svn trunk 13544. >> >> Yuriy >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f69dca62/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 10:04:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 2 Jun 2009 18:04:14 +0100 Subject: [Freeswitch-users] Outbound socket question Message-ID: Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f68fdd95/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 2 10:18:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 12:18:29 -0500 Subject: [Freeswitch-users] Outbound socket question In-Reply-To: References: Message-ID: <191c3a030906021018t5f913b49u593c286bc7324d64@mail.gmail.com> yes the socket remains open the duration of your connection. and the uuid becomes optional at that point for sendmsg but may still come into play for some FSAPI based commands like uuid_getvar On Tue, Jun 2, 2009 at 12:04 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?m going some work with outbound socket, and have a few questions. > > > > When each call is answered, I get a connection to my server socket. > > > > Is it right to assume that this connection will remain for the duration of > the call? > > > > If so, do I still need to pass the UUID when I call an application such as > playfile? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/b9436296/attachment-0002.html From dave at 3c.co.uk Tue Jun 2 10:26:30 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 2 Jun 2009 18:26:30 +0100 Subject: [Freeswitch-users] Outbound socket question References: Message-ID: <2206B39F96274B17B468DF2634FBB012@DELL9> Hi Nik, Yes and no, respectively. Cheers -- Dave ----- Original Message ----- From: Nik Middleton To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 02, 2009 6:04 PM Subject: [Freeswitch-users] Outbound socket question Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/095e94bb/attachment-0002.html From msc at freeswitch.org Tue Jun 2 11:02:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:02:53 -0700 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: Message-ID: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: > How to update FreeSWITCH-mod_sofia/1.0.3-12163?? Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible. Let us know how it goes... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/de577ef7/attachment-0002.html From msc at freeswitch.org Tue Jun 2 11:18:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:18:41 -0700 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> References: <2613803.271261243959087777.JavaMail.root@zimbra.tng.de> <27754943.271351243959529181.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906021118g18fdd313n66625e5f414d4fe8@mail.gmail.com> On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert wrote: > Hello, > > I have problems with sending DTMF. The freeswitch server recieves the > digits from my telefon (SIP account) if I activate DTMF with: > > in my dialplan. > > The line > <--param name="dtmf-type" value="rfc2833"/--> > in sofia.conf.xml is active, too. > > But than I have the problem that the other phone doesn't work. It is a > VoATM device. The curious thing is that I see the digits in the logfile > whiche were sent from the phone. In the first example I saw nothing. > > Does anybody have an idea??? > Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/dce80840/attachment-0002.html From msc at freeswitch.org Tue Jun 2 11:14:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 11:14:49 -0700 Subject: [Freeswitch-users] effective_caller_id_number on bridge dialstring In-Reply-To: <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> References: <200906021404.07496.yivzhenko@mksat.net> <191c3a030906020950mc30e4ahd1a971cb75de45e6@mail.gmail.com> Message-ID: <87f2f3b90906021114i70b12db5h62a7d021b9091eb2@mail.gmail.com> On Tue, Jun 2, 2009 at 9:50 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > effective_* is *NOT EVER* valid in the dial string. they are settings of > an existing session to control what caller id they pass. > > FYI, I've updated the wiki to reflect this fact and to make it completely obvious: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/887f0f10/attachment-0002.html From rdenert at tng.de Tue Jun 2 11:37:14 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 2 Jun 2009 20:37:14 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <14151206.271881243967650954.JavaMail.root@zimbra.tng.de> Message-ID: <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From keithl at voxtelecom.co.za Tue Jun 2 12:12:20 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Tue, 2 Jun 2009 21:12:20 +0200 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: Message-ID: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Hi, Try starting using the -nonat switch. Best Regards Keith From: Muhammad Shahzad [mailto:shaheryarkh at googlemail.com] Sent: 02 June 2009 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch taking too long to start up Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: As I understand it, a new ?feature? was added over the weekend to resolve NAT. If you?re firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/62270a27/attachment-0002.html From msc at freeswitch.org Tue Jun 2 12:29:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 12:29:19 -0700 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> References: <14151206.271881243967650954.JavaMail.root@zimbra.tng.de> <27178790.271901243967834365.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906021229v5784466dv2932aaa162c052d8@mail.gmail.com> On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert wrote: > Hello, > > I'm not sure which one is it. But I think I send the digits in RFC 2833. > All devicves are supporting RFC 2833. > Actually, that's very interesting. The start_dtmf app looks for in-band DTMFs, so if you have to have that in order to hear digits from the other end then the other end most definitely isn't doing RFC2833. For the sake of testing, try sending in-band and see how the other end reacts. Might want to check this out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate My guess is that the equipment along the way is futzing with things. FreeSWITCH does break easily but it does find bugs in other VoIP systems that it talks to... :) -MC > > The equipment: > (VoATM) > Allied Data Copperjet 1614 (ISDN) > Siemens Euroset 5020 phone > > (MGCP) > Thomson SpeedTouch 780WL > Siemens Euroset 5020 phone > > (SIP) > AVM Fritz!Box 7170 > Siemens Euroset 5020 phone > > If necessary I can send my configuration. > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Michael Collins" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 > Amsterdam/Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Prolems with DTMF > > > > > > On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: > > > Hello, > > I have problems with sending DTMF. The freeswitch server recieves the > digits from my telefon (SIP account) if I activate DTMF with: > > in my dialplan. > > The line > <--param name="dtmf-type" value="rfc2833"/--> > in sofia.conf.xml is active, too. > > But than I have the problem that the other phone doesn't work. It is a > VoATM device. The curious thing is that I see the digits in the logfile > whiche were sent from the phone. In the first example I saw nothing. > > Does anybody have an idea??? > > > Are you trying to send digits inband or RFC2833? Unless there's a > compelling reason not to, we recommend 2833. What is the equipment on the > far end looking for? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/71075867/attachment-0002.html From dome at tel.co.th Tue Jun 2 12:42:27 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 3 Jun 2009 02:42:27 +0700 Subject: [Freeswitch-users] How to change sound-path when switch language Message-ID: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> Dear sir, i create mod_say_th for Thai language. i found some problem about sound-path. I have config th.xml in conf/lang/th/ ... when i try Freeswitch still looking sounf file in /sounds/en/us/callie (en sound-path) Someone help me please Dome C. From brian at freeswitch.org Tue Jun 2 12:50:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 14:50:06 -0500 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> Message-ID: <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> You'll need to set the variable default_language /b On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote: > Dear sir, > i create mod_say_th for Thai language. i found some problem > about sound-path. > I have config th.xml in conf/lang/th/ > tts-engine="cepstral" tts-voice="callie"> > ... > > when i try > > Freeswitch still looking sounf file in /sounds/en/us/callie (en > sound-path) > > Someone help me please Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/87eb8083/attachment-0002.html From fvillarroel at yahoo.com Tue Jun 2 13:46:46 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Tue, 2 Jun 2009 13:46:46 -0700 (PDT) Subject: [Freeswitch-users] Passthru mode Message-ID: <302665.69883.qm@web34301.mail.mud.yahoo.com> Dear, I can't solve my problem, i was try with: and: in freeswitch.xml But receive the same log: http://pastebin.freeswitch.org/9204 Anyone help me. Fernando --- On Mon, 6/1/09, FERNANDO VILLARROEL wrote: > From: FERNANDO VILLARROEL > Subject: Re: [Freeswitch-users] Passthru mode > To: freeswitch-users at lists.freeswitch.org > Date: Monday, June 1, 2009, 8:10 PM > > Hello i was try with: > > > data="sofia/gateway/ubb/$1$2$3"/> > > This is the log on FS_CLI: > > http://pastebin.freeswitch.org/9204 > > Fernando > > --- On Mon, 6/1/09, Michael Collins > wrote: > > > From: Michael Collins > > Subject: Re: [Freeswitch-users] Passthru mode > > To: freeswitch-users at lists.freeswitch.org > > Date: Monday, June 1, 2009, 7:41 PM > > > > > > On Mon, Jun 1, 2009 at 3:20 PM, > > FERNANDO VILLARROEL > > wrote: > > > > > > > > Hello the dial plan: > > > > > > > > > data="sofia/gateway/ubb/$1$2$3"/> > > > > > > > > This i setup from Wikipbx. > > What about this in the dialplan? > > > data="proxy_media=true"/> > > Or alternatively this in the SIP profile? > > > > > value="true"/> > > > > I just want to make sure you're actually telling FS > to > > use proxy media. If I may make a suggestion: use > pastebin.freeswitch.org > > and pastebin the entire extension in the dialplan as > well as > > a complete debug log of the call from the FS CLI. > Please see > > this page for some handy tips on gathering information > for > > troubleshooting: > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > -MC > > > > > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ? ? ? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From larclap at yahoo.com Tue Jun 2 14:53:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 2 Jun 2009 14:53:38 -0700 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: <035b01c9e3cc$96486450$c2d92cf0$@com> Brian, I'm probably not the only one here, but much of what I have to do to get Freeswitch going is new to me. Never installed or really worked with Linux and scripting; just a little xml. It is challenging. Freeswitch is interesting, appealing and challenging. The work your group has done is amazing. Given this, interacting with you can be intimidating. I am experiencing the slow start with build 13532. I assume that "block all ICMP" refers to the firewall/gateway. If this is correct, why is it that I can ping the firewall from the Freeswitch box? Can you explain in more detail what it might be on my network that is blocking ICMP? All my clients and Freeswitch itself are behind a NAT firewall. Thanks Lars Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 01, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/bc729270/attachment-0002.html From brian at freeswitch.org Tue Jun 2 15:01:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 17:01:51 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <035b01c9e3cc$96486450$c2d92cf0$@com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: <315C26CB-8738-46E4-A2CD-AB812BBB9200@freeswitch.org> We are working to correct it. So hold on ;) /b On Jun 2, 2009, at 4:53 PM, Lars Zeb wrote: > Brian, > > I?m probably not the only one here, but much of what I have to do to > get Freeswitch going is new to me. Never installed or really worked > with Linux and scripting; just a little xml. It is challenging. > Freeswitch is interesting, appealing and challenging. The work your > group has done is amazing. Given this, interacting with you can be > intimidating. > > I am experiencing the slow start with build 13532. I assume that > ?block all ICMP? refers to the firewall/gateway. If this is correct, > why is it that I can ping the firewall from the Freeswitch box? Can > you explain in more detail what it might be on my network that is > blocking ICMP? All my clients and Freeswitch itself are behind a NAT > firewall. > > Thanks Lars > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 > i686 i386 GNU/Linux Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/f2b02bb4/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 2 15:14:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 2 Jun 2009 17:14:29 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> Message-ID: <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> You have a good point. On the other hand, it's just another random day in SVN trunk. =D Most projects don't offer SVN trunk you can play spin-the-bottle with and land on something production-ready. But we are pretty close most of the time. Here's my point of view: That particular addition was a component to the core meant to be transparent. If we did not find out the hard-way about this by adding it to trunk, we would have found out the even-harder-way by having it imprinted in the actual release. We try to keep the suffering to a minimum but we sometimes fall short. On Tue, Jun 2, 2009 at 6:23 AM, David Knell wrote: > At the risk of evisceration (but with the intention of helping avoid > future brain dead build vs. idiot admin debates), I'd suggest that, when > significant new bits are added to the switch core, they should default to > being off and require a configuration option to turn them on. Such config > options can be added to the default config; that way new installs will have > the new functionality enabled by default, but those upgrading from an older > install will need to enable them manually, reducing the risk of stuff > breaking. > > --Dave > > ----- Original Message ----- > *From:* Brian West > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, June 01, 2009 11:33 PM > *Subject:* Re: [Freeswitch-users] Make current fails (build 13537) > > NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since > your network must be eating the packets its sending out to detect if you're > behind nat or not... and not getting an ICMP unreachable like it should be > getting... the joys of admins that block all ICMP like idiots. ICMP has > many uses... and outright blocking it is stupid. (This is my assumption cuz > its what makes sense in this case) > So you're getting hit by the nice retry/timeout loop in the natpmp software > we just added and possibly the upnp lib too. > > So for now edit switch_core.c and comment out switch_nat_init(); > > I'm working my ass off to ensure that our users that do have to live in > these insane nat scenarios can do so without much if any pain. Most of which > uses SMB/Consumer grade routers which these two libs we added would allow us > to poke holes and setup stuff and make it painless as possible. > > Soon you'll have an option in switch.conf.xml to turn it off. > > Please next time don't be so demanding and calling builds brain dead .. > when in fact its trying to become more aware of its network config without > much user input. > > /b > > On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: > > Well I can only assume build 13537 is brain dead. Surely I shouldn?t > have to edit a whole bunch of configs to get it working. FS now takes 3 > minutes to start, with no indication as to what it?s looking for in the > logs. That said, to date ?make current? has always worked well for me. > Guess I was bound to hit a bad one eventually. > Still, it?s very frustrating. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/8f036afb/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Jun 2 16:37:27 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 3 Jun 2009 00:37:27 +0100 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <035b01c9e3cc$96486450$c2d92cf0$@com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: As Anthony comments later, using SVN for updates is usually a risky business for most projects. We all have been blessed by fantastic coding to date with this project, that has lulled us into believing that using the latest snapshot will be OK. This is the first time that I've had problems. I have no doubt that the DEV's have taken this onboard, but it can sometimes be a reality check to realize that the subscribed based has grown to such a size that regression testing now becomes mandatory if the project is to move onto the next stage. A very valid comment was made on this thread that new features should be disabled by default until thoroughly tested. It's all part of the learning cycle. In my view the trunk needs to be updated more frequently and this should be what us mere mortals use. To often I see messages saying you're using a 2 week old version, that bug's been fixed. FS, is coming to a level where code has to be managed in a more structured way, but I have now doubt this will be addressed fairly rapidly. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: 02 June 2009 22:54 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) Brian, I'm probably not the only one here, but much of what I have to do to get Freeswitch going is new to me. Never installed or really worked with Linux and scripting; just a little xml. It is challenging. Freeswitch is interesting, appealing and challenging. The work your group has done is amazing. Given this, interacting with you can be intimidating. I am experiencing the slow start with build 13532. I assume that "block all ICMP" refers to the firewall/gateway. If this is correct, why is it that I can ping the firewall from the Freeswitch box? Can you explain in more detail what it might be on my network that is blocking ICMP? All my clients and Freeswitch itself are behind a NAT firewall. Thanks Lars Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 01, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make current fails (build 13537) NO its not a bad one at all. Its switch_nat_init(); in switch_core.c since your network must be eating the packets its sending out to detect if you're behind nat or not... and not getting an ICMP unreachable like it should be getting... the joys of admins that block all ICMP like idiots. ICMP has many uses... and outright blocking it is stupid. (This is my assumption cuz its what makes sense in this case) So you're getting hit by the nice retry/timeout loop in the natpmp software we just added and possibly the upnp lib too. So for now edit switch_core.c and comment out switch_nat_init(); I'm working my ass off to ensure that our users that do have to live in these insane nat scenarios can do so without much if any pain. Most of which uses SMB/Consumer grade routers which these two libs we added would allow us to poke holes and setup stuff and make it painless as possible. Soon you'll have an option in switch.conf.xml to turn it off. Please next time don't be so demanding and calling builds brain dead .. when in fact its trying to become more aware of its network config without much user input. /b On Jun 1, 2009, at 5:24 PM, Nik Middleton wrote: Well I can only assume build 13537 is brain dead. Surely I shouldn't have to edit a whole bunch of configs to get it working. FS now takes 3 minutes to start, with no indication as to what it's looking for in the logs. That said, to date 'make current' has always worked well for me. Guess I was bound to hit a bad one eventually. Still, it's very frustrating. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/d651d2f1/attachment-0002.html From msc at freeswitch.org Tue Jun 2 16:38:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 16:38:21 -0700 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <191c3a030906021514v51857e64w49796f5dc3877d4c@mail.gmail.com> Message-ID: <87f2f3b90906021638t51d3dfcdo40426d2f13c22ffc@mail.gmail.com> On Tue, Jun 2, 2009 at 3:14 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You have a good point. > > On the other hand, it's just another random day in SVN trunk. =D > Most projects don't offer SVN trunk you can play spin-the-bottle with and > land on something production-ready. But we are pretty close most of the > time. > > Here's my point of view: > That particular addition was a component to the core meant to be > transparent. > If we did not find out the hard-way about this by adding it to trunk, > we would have found out the even-harder-way by having it imprinted in the > actual release. > > We try to keep the suffering to a minimum but we sometimes fall short. > This is also why we need as many people as possible updating FS as often as possible. The greater the number of environments we have running FreeSWITCH, the less likely it is that stuff like this will sneak through and the more likely it will be caught and fixed quickly. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/a273254f/attachment-0002.html From brian at freeswitch.org Tue Jun 2 16:46:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 2 Jun 2009 18:46:10 -0500 Subject: [Freeswitch-users] Make current fails (build 13537) In-Reply-To: References: <5052B519-AE5B-47A4-BC6A-0AE40C2ED882@freeswitch.org> <1998CC8A-E6B4-454C-8BB4-E4FDB6195733@freeswitch.org> <035b01c9e3cc$96486450$c2d92cf0$@com> Message-ID: You now have -nonat and the hang on start up with the nat detection code is fixed now. /b On Jun 2, 2009, at 6:37 PM, Nik Middleton wrote: > As Anthony comments later, using SVN for updates is usually a risky > business for most projects. We all have been blessed by fantastic > coding to date with this project, that has lulled us into believing > that using the latest snapshot will be OK. This is the first time > that I?ve had problems. > > I have no doubt that the DEV?s have taken this onboard, but it can > sometimes be a reality check to realize that the subscribed based > has grown to such a size that regression testing now becomes > mandatory if the project is to move onto the next stage. > > A very valid comment was made on this thread that new features > should be disabled by default until thoroughly tested. It?s all part > of the learning cycle. In my view the trunk needs to be updated > more frequently and this should be what us mere mortals use. To > often I see messages saying you?re using a 2 week old version, that > bug?s been fixed. > > FS, is coming to a level where code has to be managed in a more > structured way, but I have now doubt this will be addressed fairly > rapidly. > > Regards, > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/40b69769/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 17:41:52 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 08:41:52 +0800 Subject: [Freeswitch-users] How to reload xml without using console command line?? In-Reply-To: References: Message-ID: Thanks a lot ! This's what i want. 2009/6/2 Michael Jerris > /usr/local/freeswitch/bin/fs_cli -x reloadxml > > On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote: > > > How to reload xml without using console command line?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/e2f62f36/attachment-0002.html From dujinfang at gmail.com Tue Jun 2 19:20:43 2009 From: dujinfang at gmail.com (seven) Date: Wed, 3 Jun 2009 10:20:43 +0800 Subject: [Freeswitch-users] Is there a way to cancel att_xfer? Message-ID: <0FC73ACB-C36D-4CE3-A2C6-7E3CB9AD63C8@gmail.com> Hi, Assume the following sinario: A call B, B att_xfer to C if no answer on C for a long time, B can cancel the att_xfer by pressing a key and talk to A again. Is that possible? Thank you. 7. From brad.tuan at gmail.com Tue Jun 2 19:49:15 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 10:49:15 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: I was updated my FS and rebuilt it. It works........ But when User2(FS2) accept the call from User1(FS1) , User2(FS2) display "call established",but User1(FS1) still display "calling". Why?? (I think maybe that I need to do some setting on FS2.) 2009/6/2 Brad Tuan > FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 > > These two FS are in the same LAN. > > I just try to pass one sip call from one FS to another. > > If it works, next is FS1( PublicIP ) to FS2( PublicIP ). > > 2009/6/2 Ken Rice > > Dumb question... Is 187 the local fs machine? You should have the IP >> address of the remote FS machine >> >> >> ------------------------------ >> *From: *Brad Tuan >> *Reply-To: * >> *Date: *Tue, 2 Jun 2009 17:00:29 +0800 >> *To: * >> *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to >> another FS ?? >> >> >> the same message........ >> >> 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 100 >> 1->97710001 in context default >> 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() >> Cannot l >> ocate registered user 97710001 at 192.168.141.187 at internal >> 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() >> Close Cha >> nnel N/A [CS_NEW] >> 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 >> switch_ivr_originate() Can >> not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() >> Originate >> Failed. Cause: USER_NOT_REGISTERED >> >> 2009/6/2 Jason White >> >> Brad Tuan wrote: >> > I have tried >> > >> > >> > >> > > data="sofia/internal/$1%192.168.141.187"/> >> >> Change the % to an @ in the above. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/a8416abf/attachment-0002.html From plite2012 at gmail.com Tue Jun 2 20:46:16 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 2 Jun 2009 22:46:16 -0500 Subject: [Freeswitch-users] Sending fax on Windows: Any one has succeeded? Message-ID: Has anyone succeeded in sending fax on Windows with the following command line? originate sofia/gateway// &txfax(/path_to_fax_file) No matter how I specify that path (I even copied the file into the installation folder, C:\Program Files\FreeSWITCH), I always got "[ERR] mod_fax.c:518 process_fax() Cannot send inexistant fax file". Any hint would be highly appreciated! From brad.tuan at gmail.com Tue Jun 2 21:38:56 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 12:38:56 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: ......I've update my FS by SVN.......... but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" Is that right?? And the displayname is still "97730002"....... What i confused is why "97730002" ?? ( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) ) recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: ------------------------------------------------------------------------ INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 ;tag=124393762732run00 To: Contact: Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: From: 97719006 ;tag=124393762732run00 To: Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006 at 61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: ------------------------------------------------------------------------ INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" >;tag=F9rteQHjgS52m To: Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" >;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2009/6/3 Michael Collins > > > On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: > >> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? > > > Your best bet is to use SVN trunk. It is the most stable version available, > even more stable than the latest 1.0.4pre8 release candidate. Back up your > entire freeswitch folder in case there's an issue. Hopefully you're running > in Linux, so you could do: > mv /usr/local/freeswitch /usr/local/freeswitch.bak > > Then use the quick and dirty install from the wiki: > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > When the install is finished you will have a fresh copy of FS and a brand > new default configuration. You'll need to go back and enable and build any > modules you need that aren't done by default. You will also need to re-apply > any changes you made to the default configuration from your previous > install. Hopefully you didn't have to edit any of the files or maybe just a > few, like vars.xml. In any case, I recommend editing as few of the default > config files as possible. > > Let us know how it goes... > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/df2c45e6/attachment-0002.html From msc at freeswitch.org Tue Jun 2 21:59:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 2 Jun 2009 21:59:10 -0700 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> okay, you will need to use pastebin and post your configuration. anything you changed from the default config, especially in the dialplan, but also vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console loglevel 7") and also do the SIP trace. Make a few test calls and capture all the output. -MC On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > ......I've update my FS by SVN.......... > > but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" > > Is that right?? > > And the displayname is still "97730002"....... > > What i confused is why "97730002" ?? > > ( I have users from 97730000~97739999,but when I call them from 97710006 , > the display name is always "97730002"(it should be "97710006".....) ) > > recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: > ------------------------------------------------------------------------ > INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 > Via: SIP/2.0/UDP 61.61.162.130:5060 > ;branch=z9hG4bKrun12440031628377850000 > Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 > From: 97719006 ;tag=124393762732run00 > To: > Contact: > Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 > CSeq: 8500 INVITE > Max-Forwards: 68 > Content-Type: application/sdp > Content-Length: 237 > v=0 > o=169 0 0 IN IP4 61.61.162.130 > s=ots > c=IN IP4 61.61.162.130 > t=0 0 > m=audio 5158 RTP/AVP 18 8 0 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:18 G729/8000/1 > a=rtpmap:101 telephone-event/8000/1 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 > Via: SIP/2.0/UDP 61.61.162.130:5060 > ;branch=z9hG4bKrun12440031628377850000 > Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 > Record-Route: > From: 97719006 ;tag=124393762732run00 > To: > Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 > CSeq: 8500 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New > Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 > f34cc3da] > 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 977 > 19006->97730009 in context default > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 1 execute_extension::dx XML features > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 2 > record_session::C:\SipGo/recordings/97719006.2009-06-03-12 > -25-59.wav > 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 > switch_ivr_bind_dtmf_meta_ses > sion() Bound B-Leg: 3 execute_extension::cf XML features > 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New > Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 > ;rinstance=89358e5ea9aaa > 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] > 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 > switch_ivr_originate() Se > nding early media > 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring > SDP: > v=0 > o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 > s=FreeSWITCH > c=IN IP4 203.64.215.209 > t=0 0 > m=audio 17022 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() > Pre-Answer > sofia/internal/97719006 at 61.61.162.130:5060! > send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: > ------------------------------------------------------------------------ > INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e > SIP/2.0 > Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta > Max-Forwards: 67 > From: "Extension 97730002" > >;tag=F9rteQHjgS52m > To: > Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace > CSeq: 115883243 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-descrip > tion, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 449 > Remote-Party-ID: "Extension 97730002" > >;party=cal > ling;screen=yes;privacy=off > v=0 > o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 > 203.64.215.209 > s=FreeSWITCH > c=IN IP4 203.64.215.209 > t=0 0 > m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 > a=rtpmap:8 PCMA/8000 > a=rtpmap:115 G7221/32000 > a=fmtp:115 bitrate=48000 > a=rtpmap:107 G7221/16000 > a=fmtp:107 bitrate=32000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > 2009/6/3 Michael Collins > >> >> >> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >> >>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >> >> >> Your best bet is to use SVN trunk. It is the most stable version >> available, even more stable than the latest 1.0.4pre8 release candidate. >> Back up your entire freeswitch folder in case there's an issue. Hopefully >> you're running in Linux, so you could do: >> mv /usr/local/freeswitch /usr/local/freeswitch.bak >> >> Then use the quick and dirty install from the wiki: >> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >> >> When the install is finished you will have a fresh copy of FS and a brand >> new default configuration. You'll need to go back and enable and build any >> modules you need that aren't done by default. You will also need to re-apply >> any changes you made to the default configuration from your previous >> install. Hopefully you didn't have to edit any of the files or maybe just a >> few, like vars.xml. In any case, I recommend editing as few of the default >> config files as possible. >> >> Let us know how it goes... >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090602/1a1e43d3/attachment-0002.html From shaheryarkh at googlemail.com Tue Jun 2 22:09:22 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 3 Jun 2009 11:09:22 +0600 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Message-ID: I had to upgrade again svn revision to use this switch, but it works. Thank you. On Wed, Jun 3, 2009 at 1:12 AM, Keith Laaks wrote: > Hi, > > > > Try starting using the -nonat switch. > > > > Best Regards > > > > Keith > > > > *From:* Muhammad Shahzad [mailto:shaheryarkh at googlemail.com] > *Sent:* 02 June 2009 14:39 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch taking too long to start up > > > > Yes, this resolves the problem. > > Thank you. > > On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: > > Actually Brain mentioned that you can comment out switch_nat_init(); in > switch_core.c > > > > On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: > > As I understand it, a new ?feature? was added over the weekend to > resolve NAT. If you?re firewall is not allowing ICMP then FS waits until it > times out. At this time there is no option to disable it. > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Muhammad Shahzad > *Sent:* 02 June 2009 11:40 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch taking too long to start up > > > > Hi, > > I have just upgraded Freeswitch from svn revision 12432 to 13544. I am > using 32bit CentOS 5.3, "make current" command completes successfully > without any errors but when i start freeswitch it take considerable time > (roughly 90 - 120 seconds) to start up. During this time no message is > display on console. Once successfully started, it works fine. However, this > initial delay is really annoying. Is there anyway to reduce/remove this > delay? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/00066170/attachment-0002.html From brad.tuan at gmail.com Tue Jun 2 22:26:42 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 13:26:42 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> Message-ID: I only change freeSWITCH\conf\dialplan\default.xml and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default 2009/6/3 Michael Collins > okay, you will need to use pastebin and post your configuration. anything > you changed from the default config, especially in the dialplan, but also > vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console > loglevel 7") and also do the SIP trace. Make a few test calls and capture > all the output. > > -MC > > > On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > >> ......I've update my FS by SVN.......... >> >> but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" >> >> Is that right?? >> >> And the displayname is still "97730002"....... >> >> What i confused is why "97730002" ?? >> >> ( I have users from 97730000~97739999,but when I call them from 97710006 , >> the display name is always "97730002"(it should be "97710006".....) ) >> >> recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 >> Record-Route: >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> From: 97719006 ;tag=124393762732run00 >> To: >> Contact: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> Max-Forwards: 68 >> Content-Type: application/sdp >> Content-Length: 237 >> v=0 >> o=169 0 0 IN IP4 61.61.162.130 >> s=ots >> c=IN IP4 61.61.162.130 >> t=0 0 >> m=audio 5158 RTP/AVP 18 8 0 101 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=rtpmap:101 telephone-event/8000/1 >> a=fmtp:101 0-15 >> >> ------------------------------------------------------------------------ >> send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> Record-Route: >> From: 97719006 ;tag=124393762732run00 >> To: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 >> f34cc3da] >> 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 977 >> 19006->97730009 in context default >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 1 execute_extension::dx XML features >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 2 >> record_session::C:\SipGo/recordings/97719006.2009-06-03-12 >> -25-59.wav >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 3 execute_extension::cf XML features >> 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 >> ;rinstance=89358e5ea9aaa >> 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] >> 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 >> switch_ivr_originate() Se >> nding early media >> 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring >> SDP: >> v=0 >> o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 17022 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() >> Pre-Answer >> sofia/internal/97719006 at 61.61.162.130:5060! >> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e >> SIP/2.0 >> Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta >> Max-Forwards: 67 >> From: "Extension 97730002" >> >;tag=F9rteQHjgS52m >> To: >> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace >> CSeq: 115883243 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-descrip >> tion, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 449 >> Remote-Party-ID: "Extension 97730002" >> >;party=cal >> ling;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 >> 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:115 G7221/32000 >> a=fmtp:115 bitrate=48000 >> a=rtpmap:107 G7221/16000 >> a=fmtp:107 bitrate=32000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> 2009/6/3 Michael Collins >> >>> >>> >>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >>> >>>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >>> >>> >>> Your best bet is to use SVN trunk. It is the most stable version >>> available, even more stable than the latest 1.0.4pre8 release candidate. >>> Back up your entire freeswitch folder in case there's an issue. Hopefully >>> you're running in Linux, so you could do: >>> mv /usr/local/freeswitch /usr/local/freeswitch.bak >>> >>> Then use the quick and dirty install from the wiki: >>> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >>> >>> When the install is finished you will have a fresh copy of FS and a brand >>> new default configuration. You'll need to go back and enable and build any >>> modules you need that aren't done by default. You will also need to re-apply >>> any changes you made to the default configuration from your previous >>> install. Hopefully you didn't have to edit any of the files or maybe just a >>> few, like vars.xml. In any case, I recommend editing as few of the default >>> config files as possible. >>> >>> Let us know how it goes... >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/3c0f510e/attachment-0002.html From woodydickson at gmail.com Tue Jun 2 23:22:11 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Wed, 3 Jun 2009 14:22:11 +0800 Subject: [Freeswitch-users] Set problem in dialplan Message-ID: Hello, I am getting a strange problem in my dialplan. After doing "SET", I want to use it in the next condition field. But then the value is not being set properly. Could someone please tell me what is wrong? Thanks, Woody Here is the dialplan: Here is the FS log. Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution->get-pin] continue=true Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [get-pin] ${destination_number}(117) =~ /^(.*)$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Action set(conference_id=111) Dialplan: sofia/internal/1001 at 192.168.1.101 Action set(is_moderator=true) Dialplan: sofia/internal/1001 at 192.168.1.101 Action info() Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution->conf] continue=false Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] ${is_moderator}() =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] ${is_moderator}() =~ /^false$/ break=never Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [conf] ${is_moderator}() =~ /^$/ break=always Dialplan: sofia/internal/1001 at 192.168.1.101 Action playback(/var/app/prompt/wav/bye.wav) Dialplan: sofia/internal/1001 at 192.168.1.101 Action hangup() -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/f6b6ec9b/attachment-0002.html From mrene_lists at avgs.ca Tue Jun 2 23:26:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 02:26:32 -0400 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: Message-ID: <63B78532-2DF8-47E2-91CF-060AF95B8205@avgs.ca> Hi, FreeSWITCH decides what to execute first, the set application runs later (look a few lines later, you'll see lines beginning with EXECUTE, this is when it runs). If you need to use variables you've set in the DP, you need to use the transfer application to make it go back into routing state. Math On 3-Jun-09, at 2:22 AM, Woody Dickson wrote: > Hello, > > I am getting a strange problem in my dialplan. > > After doing "SET", I want to use it in the next condition field. > But then the value is not being set properly. > > Could someone please tell me what is wrong? > > Thanks, > Woody > > > Here is the dialplan: > > > break="never"> > > > > > > > > > > > > > > > > > > > > > > > > > > > Here is the FS log. > > Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution- > >get-pin] continue=true > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [get-pin] $ > {destination_number}(117) =~ /^(.*)$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Action > set(conference_id=111) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action > set(is_moderator=true) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action info() > Dialplan: sofia/internal/1001 at 192.168.1.101 parsing [conf-execution- > >conf] continue=false > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] $ > {is_moderator}() =~ /^true$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (FAIL) [conf] $ > {is_moderator}() =~ /^false$/ break=never > Dialplan: sofia/internal/1001 at 192.168.1.101 Regex (PASS) [conf] $ > {is_moderator}() =~ /^$/ break=always > Dialplan: sofia/internal/1001 at 192.168.1.101 Action playback(/var/app/ > prompt/wav/bye.wav) > Dialplan: sofia/internal/1001 at 192.168.1.101 Action hangup() > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/1efcaa0a/attachment-0002.html From jim at evolutiontel.net Tue Jun 2 23:32:49 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 3 Jun 2009 16:32:49 +1000 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <302665.69883.qm@web34301.mail.mud.yahoo.com> References: <302665.69883.qm@web34301.mail.mud.yahoo.com> Message-ID: Fernando, Try setting 'inbound-late-negotiation' in your SIP Profile. This will allow the call to hit the dialplan where you can set proxy_media. This also assumes you have bypass_media set to false in your dialplan. Alternatively I beleive you can set "inbound-proxy-media" in the SIP Profile and this will do the same thing. Regards, Jim On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL wrote: > > Dear, > > I can't solve my problem, i was try with: > > > > > and: > > in freeswitch.xml > > But receive the same log: > > http://pastebin.freeswitch.org/9204 > > Anyone help me. > > Fernando > > --- On Mon, 6/1/09, FERNANDO VILLARROEL wrote: > >> From: FERNANDO VILLARROEL >> Subject: Re: [Freeswitch-users] Passthru mode >> To: freeswitch-users at lists.freeswitch.org >> Date: Monday, June 1, 2009, 8:10 PM >> >> Hello i was try with: >> >> >> > data="sofia/gateway/ubb/$1$2$3"/> >> >> This is the log on FS_CLI: >> >> http://pastebin.freeswitch.org/9204 >> >> Fernando >> >> --- On Mon, 6/1/09, Michael Collins >> wrote: >> >> > From: Michael Collins >> > Subject: Re: [Freeswitch-users] Passthru mode >> > To: freeswitch-users at lists.freeswitch.org >> > Date: Monday, June 1, 2009, 7:41 PM >> > >> > >> > On Mon, Jun 1, 2009 at 3:20 PM, >> > FERNANDO VILLARROEL >> > wrote: >> > >> > >> > >> > Hello the dial plan: >> > >> > >> > >> > > > data="sofia/gateway/ubb/$1$2$3"/> >> > >> > >> > >> > This i setup from Wikipbx. >> > What about this in the dialplan? >> > > > data="proxy_media=true"/> >> > Or alternatively this in the SIP profile? >> > >> > > > value="true"/> >> > >> > I just want to make sure you're actually telling FS >> to >> > use proxy media. If I may make a suggestion: use >> pastebin.freeswitch.org >> > and pastebin the entire extension in the dialplan as >> well as >> > a complete debug log of the call from the FS CLI. >> Please see >> > this page for some handy tips on gathering information >> for >> > troubleshooting: >> > >> > http://wiki.freeswitch.org/wiki/Reporting_Bugs >> > >> > -MC >> > >> > >> > >> > -----Inline Attachment Follows----- >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mrene_lists at avgs.ca Tue Jun 2 23:33:45 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 02:33:45 -0400 Subject: [Freeswitch-users] Passthru mode In-Reply-To: References: <302665.69883.qm@web34301.mail.mud.yahoo.com> Message-ID: <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> On 3-Jun-09, at 2:32 AM, Jim Burke wrote: > Fernando, > > Try setting 'inbound-late-negotiation' in your SIP Profile. This will > allow the call to hit the dialplan where you can set proxy_media. > This also assumes you have bypass_media set to false in your dialplan. > > Alternatively I beleive you can set "inbound-proxy-media" in the SIP > Profile and this will do the same thing. But you still need late negotiation for that to work, so in both cases you need to fix that :D Math > > > Regards, > Jim > > On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL > wrote: >> >> Dear, >> >> I can't solve my problem, i was try with: >> >> >> >> >> and: >> >> in freeswitch.xml >> >> But receive the same log: >> >> http://pastebin.freeswitch.org/9204 >> >> Anyone help me. >> >> Fernando >> >> --- On Mon, 6/1/09, FERNANDO VILLARROEL >> wrote: >> >>> From: FERNANDO VILLARROEL >>> Subject: Re: [Freeswitch-users] Passthru mode >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Monday, June 1, 2009, 8:10 PM >>> >>> Hello i was try with: >>> >>> >>> >> data="sofia/gateway/ubb/$1$2$3"/> >>> >>> This is the log on FS_CLI: >>> >>> http://pastebin.freeswitch.org/9204 >>> >>> Fernando >>> >>> --- On Mon, 6/1/09, Michael Collins >>> wrote: >>> >>>> From: Michael Collins >>>> Subject: Re: [Freeswitch-users] Passthru mode >>>> To: freeswitch-users at lists.freeswitch.org >>>> Date: Monday, June 1, 2009, 7:41 PM >>>> >>>> >>>> On Mon, Jun 1, 2009 at 3:20 PM, >>>> FERNANDO VILLARROEL >>>> wrote: >>>> >>>> >>>> >>>> Hello the dial plan: >>>> >>>> >>>> >>>> >>> data="sofia/gateway/ubb/$1$2$3"/> >>>> >>>> >>>> >>>> This i setup from Wikipbx. >>>> What about this in the dialplan? >>>> >>> data="proxy_media=true"/> >>>> Or alternatively this in the SIP profile? >>>> >>>> >>> value="true"/> >>>> >>>> I just want to make sure you're actually telling FS >>> to >>>> use proxy media. If I may make a suggestion: use >>> pastebin.freeswitch.org >>>> and pastebin the entire extension in the dialplan as >>> well as >>>> a complete debug log of the call from the FS CLI. >>> Please see >>>> this page for some handy tips on gathering information >>> for >>>> troubleshooting: >>>> >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> -MC >>>> >>>> >>>> >>>> -----Inline Attachment Follows----- >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Tue Jun 2 23:39:09 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 3 Jun 2009 16:39:09 +1000 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: Message-ID: <20090603063909.GA19487@jdc.jasonjgw.net> Woody Dickson wrote: > I am getting a strange problem in my dialplan. > > After doing "SET", I want to use it in the next condition field. But then > the value is not being set properly. When parsing the dial plan, FreeSWITCH tests all of the conditions, then builds a linked list of actions to execute. Once this is done, the actions are executed, in order. This is why you can't simply set a variable in one extension and test it in the condition of a later extension. From bruce.mcalister at blueface.ie Wed Jun 3 00:16:01 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 08:16:01 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" Message-ID: <4A262331.2080104@blueface.ie> Hi All, I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the build fails with the following error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" cc: acomp failed for locks/unix/thread_mutex.c make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 I have the JDS-CBE build environment setup as recommended on the wiki with Sun Studio 12 compiler suite installed as well. I have tried building using Sun's compiler and GNU compiler but I get the same error message. I just recently tried "bootstrap.sh" prior to "configure", "make" but the error is still the same. Would someone have any suggestions for me to try to get around this? Thanks Bruce From bruce.mcalister at blueface.ie Wed Jun 3 02:47:43 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 10:47:43 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <4A262331.2080104@blueface.ie> References: <4A262331.2080104@blueface.ie> Message-ID: <4A2646BF.7090607@blueface.ie> Hi All, I get past this initial error if I change my C compiler from "usr/bin/cc" to "/usr/bin/c99". After changing the above, the compilation goes further, but I am now faced with a different error: --- /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/bin/c99 -m32 -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o threadproc/unix/signals.lo -c threadproc/unix/signals.c && touch threadproc/unix/signals.lo "threadproc/unix/signals.c", line 48: warning: implicit function declaration: kill "threadproc/unix/signals.c", line 76: incomplete struct/union/enum sigaction: act "threadproc/unix/signals.c", line 78: undefined struct/union member: sa_handler "threadproc/unix/signals.c", line 78: warning: improper pointer/integer combination: op "=" "threadproc/unix/signals.c", line 79: warning: implicit function declaration: sigemptyset "threadproc/unix/signals.c", line 79: undefined struct/union member: sa_mask "threadproc/unix/signals.c", line 80: undefined struct/union member: sa_flags "threadproc/unix/signals.c", line 103: warning: implicit function declaration: sigaction "threadproc/unix/signals.c", line 105: improper member use: sa_handler "threadproc/unix/signals.c", line 105: warning: improper pointer/integer combination: op "=" "threadproc/unix/signals.c", line 277: warning: implicit function declaration: sigdelset "threadproc/unix/signals.c", line 327: warning: implicit function declaration: sigfillset "threadproc/unix/signals.c", line 424: warning: implicit function declaration: pthread_sigmask "threadproc/unix/signals.c", line 443: warning: implicit function declaration: sigaddset c99: acomp failed for threadproc/unix/signals.c make[2]: *** [threadproc/unix/signals.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 --- Any hints towards a solution would be appreciated. Thanks Bruce Bruce McAlister wrote: > Hi All, > > I am trying to build FS 1.0.4pre8 for Solaris 10 (Update 5), however the > build fails with the following error: > > /bin/bash > /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool > --silent --mode=compile /usr/bin/cc -g -DHAVE_CONFIG_H -DSOLARIS2=10 > -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE > -I./include > -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix > -I./include/arch/unix > -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include > -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch > locks/unix/thread_mutex.lo > "/usr/include/sys/feature_tests.h", line 336: #error: "Compiler or > options invalid; UNIX 03 and POSIX.1-2001 applications require the > use of c99" > cc: acomp failed for locks/unix/thread_mutex.c > make[2]: *** [locks/unix/thread_mutex.lo] Error 1 > make[2]: Leaving directory > `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory > `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' > make: *** [libs/apr/libapr-1.la] Error 2 > From jason at jasonjgw.net Wed Jun 3 03:17:46 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 3 Jun 2009 20:17:46 +1000 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <4A2646BF.7090607@blueface.ie> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> Message-ID: <20090603101746.GA3854@jdc.jasonjgw.net> Bruce McAlister wrote: > I get past this initial error if I change my C compiler from > "usr/bin/cc" to "/usr/bin/c99". > > After changing the above, the compilation goes further, but I am now > faced with a different error: Have you tried compiling with gcc? I would also suggest starting the build procedure from the beginning. From brad.tuan at gmail.com Wed Jun 3 04:16:16 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 19:16:16 +0800 Subject: [Freeswitch-users] Little problem of group(callgroup) Message-ID: What is the ${callgroup} mean?? Is this?? >> Or this?? >>81+[group] - Add this extension to calling group #[group] (can be two digits 00-99). A beep tone confirms the function worked. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/40ed8ef0/attachment-0002.html From rdenert at tng.de Wed Jun 3 04:18:25 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 13:18:25 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <28083808.277891244027846976.JavaMail.root@zimbra.tng.de> Message-ID: <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> Hello, I have a question about the RTP stream. I made two calls with different devices. One had no problems, the other call made some difficulties. I put an extraction of my traces in attachment. (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) The first one had no problem. When I called the freeSWITCH I had a bidirectional RTP stream. I heard the announcement of the server. I could transmit digits from my telephone to the freeSWITCH which were verified by the machine. (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 phone) The second call had only a oneway audio stream. When I called the freeSWITCH I heard no announcement of the server which should be actually played. There was only a background noise. I made traces from this example. My suggestion is: In packet 378 the freeSWITCH server wants to sent RTP packets to an suspicious IP. There are over 5 packets in number. Not till then there is the correct destination IP (see packet 386). But this could be the fact that the freeSWITCH produces an error. Does anybody have an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. 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Name: traces_from 1616_to_freeSWITCH.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/19fd92f1/attachment-0002.txt From bruce.mcalister at blueface.ie Wed Jun 3 04:36:05 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 03 Jun 2009 12:36:05 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <20090603101746.GA3854@jdc.jasonjgw.net> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> <20090603101746.GA3854@jdc.jasonjgw.net> Message-ID: <4A266025.1070505@blueface.ie> Hi Jason, If I try to compile with GCC, then I am faced with the original problem where the error returns saying I need to use a c99 compatible compiler, here is the specific error: /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/libtool --silent --mode=compile /usr/sfw/bin/gcc -m32 -DHAVE_CONFIG_H -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT -D_LARGEFILE64_SOURCE -I./include -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix -I./include/arch/unix -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch locks/unix/thread_mutex.lo In file included from /usr/sfw/lib/gcc/i386-pc-solaris2.10/3.4.3/include/sys/types.h:27, from ./include/apr.h:113, from /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/include/arch/unix/apr_arch_thread_mutex.h:24, from locks/unix/thread_mutex.c:17: /usr/include/sys/feature_tests.h:336:2: #error "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" make[2]: *** [locks/unix/thread_mutex.lo] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 In all cases I have started the build from the beginning, whereby I remove and re-extract the 1.0.4pre8 tarball. I've tried with just a configure and also a bootstrap/configure, but I end up with the same error (except when I change the compiler to Sun Studio 12's c99). Is GCC 3.4.3 too old to use to build this version of freeswitch? Thanks Bruce Jason White wrote: > Bruce McAlister wrote: >> I get past this initial error if I change my C compiler from >> "usr/bin/cc" to "/usr/bin/c99". >> >> After changing the above, the compilation goes further, but I am now >> faced with a different error: > > Have you tried compiling with gcc? I would also suggest starting the build > procedure from the beginning. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brad.tuan at gmail.com Wed Jun 3 06:04:33 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 21:04:33 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <87f2f3b90906022159l51c9787csab6a6e7f0692839e@mail.gmail.com> Message-ID: I only change freeSWITCH\conf\dialplan\default.xml and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default 2009/6/3 Michael Collins > okay, you will need to use pastebin and post your configuration. anything > you changed from the default config, especially in the dialplan, but also > vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console > loglevel 7") and also do the SIP trace. Make a few test calls and capture > all the output. > > -MC > > > On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan wrote: > >> ......I've update my FS by SVN.......... >> >> but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" >> >> Is that right?? >> >> And the displayname is still "97730002"....... >> >> What i confused is why "97730002" ?? >> >> ( I have users from 97730000~97739999,but when I call them from 97710006 , >> the display name is always "97730002"(it should be "97710006".....) ) >> >> recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 203.64.215.209:5060;transport=udp SIP/2.0 >> Record-Route: >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> From: 97719006 ;tag=124393762732run00 >> To: >> Contact: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> Max-Forwards: 68 >> Content-Type: application/sdp >> Content-Length: 237 >> v=0 >> o=169 0 0 IN IP4 61.61.162.130 >> s=ots >> c=IN IP4 61.61.162.130 >> t=0 0 >> m=audio 5158 RTP/AVP 18 8 0 101 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=rtpmap:101 telephone-event/8000/1 >> a=fmtp:101 0-15 >> >> ------------------------------------------------------------------------ >> send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 >> Via: SIP/2.0/UDP 61.61.162.130:5060 >> ;branch=z9hG4bKrun12440031628377850000 >> Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 >> Record-Route: >> From: 97719006 ;tag=124393762732run00 >> To: >> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 >> CSeq: 8500 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/97719006 at 61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 >> f34cc3da] >> 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 977 >> 19006->97730009 in context default >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 1 execute_extension::dx XML features >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 2 >> record_session::C:\SipGo/recordings/97719006.2009-06-03-12 >> -25-59.wav >> 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 >> switch_ivr_bind_dtmf_meta_ses >> sion() Bound B-Leg: 3 execute_extension::cf XML features >> 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 >> switch_channel_set_name() New >> Channel sofia/internal/sip:97730009 at 210.68.184.192:62807 >> ;rinstance=89358e5ea9aaa >> 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] >> 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 >> switch_ivr_originate() Se >> nding early media >> 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring >> SDP: >> v=0 >> o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 17022 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() >> Pre-Answer >> sofia/internal/97719006 at 61.61.162.130:5060! >> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: >> >> ------------------------------------------------------------------------ >> INVITE sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e >> SIP/2.0 >> Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta >> Max-Forwards: 67 >> From: "Extension 97730002" >> >;tag=F9rteQHjgS52m >> To: >> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace >> CSeq: 115883243 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-descrip >> tion, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 449 >> Remote-Party-ID: "Extension 97730002" >> >;party=cal >> ling;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 >> 203.64.215.209 >> s=FreeSWITCH >> c=IN IP4 203.64.215.209 >> t=0 0 >> m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:115 G7221/32000 >> a=fmtp:115 bitrate=48000 >> a=rtpmap:107 G7221/16000 >> a=fmtp:107 bitrate=32000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> 2009/6/3 Michael Collins >> >>> >>> >>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote: >>> >>>> How to update FreeSWITCH-mod_sofia/1.0.3-12163?? >>> >>> >>> Your best bet is to use SVN trunk. It is the most stable version >>> available, even more stable than the latest 1.0.4pre8 release candidate. >>> Back up your entire freeswitch folder in case there's an issue. Hopefully >>> you're running in Linux, so you could do: >>> mv /usr/local/freeswitch /usr/local/freeswitch.bak >>> >>> Then use the quick and dirty install from the wiki: >>> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install >>> >>> When the install is finished you will have a fresh copy of FS and a brand >>> new default configuration. You'll need to go back and enable and build any >>> modules you need that aren't done by default. You will also need to re-apply >>> any changes you made to the default configuration from your previous >>> install. Hopefully you didn't have to edit any of the files or maybe just a >>> few, like vars.xml. In any case, I recommend editing as few of the default >>> config files as possible. >>> >>> Let us know how it goes... >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/49e162f1/attachment-0002.html From brad.tuan at gmail.com Wed Jun 3 06:06:03 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 3 Jun 2009 21:06:03 +0800 Subject: [Freeswitch-users] How to pass a call from one FS to another FS ?? In-Reply-To: References: Message-ID: I was updated my FS and rebuilt it. It works........ But when User2(FS2) accept the call from User1(FS1) , User2(FS2) display "call established",but User1(FS1) still display "calling". Why?? (I think maybe that I need to do some setting on FS2.) 2009/6/2 Brad Tuan > FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187 > > These two FS are in the same LAN. > > I just try to pass one sip call from one FS to another. > > If it works, next is FS1( PublicIP ) to FS2( PublicIP ). > > 2009/6/2 Ken Rice > > Dumb question... Is 187 the local fs machine? You should have the IP >> address of the remote FS machine >> >> >> ------------------------------ >> *From: *Brad Tuan >> *Reply-To: * >> *Date: *Tue, 2 Jun 2009 17:00:29 +0800 >> *To: * >> *Subject: *Re: [Freeswitch-users] How to pass a call from one FS to >> another FS ?? >> >> >> the same message........ >> >> 2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 100 >> 1->97710001 in context default >> 2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel() >> Cannot l >> ocate registered user 97710001 at 192.168.141.187 at internal >> 2009-06-02 16:49:01 [NOTICE] mod_sofia.c:2642 sofia_outgoing_channel() >> Close Cha >> nnel N/A [CS_NEW] >> 2009-06-02 16:49:01 [ERR] switch_ivr_originate.c:1425 >> switch_ivr_originate() Can >> not create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> 2009-06-02 16:49:01 [INFO] mod_dptools.c:1998 audio_bridge_function() >> Originate >> Failed. Cause: USER_NOT_REGISTERED >> >> 2009/6/2 Jason White >> >> Brad Tuan wrote: >> > I have tried >> > >> > >> > >> > > data="sofia/internal/$1%192.168.141.187"/> >> >> Change the % to an @ in the above. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/e0c80105/attachment-0002.html From brian at freeswitch.org Wed Jun 3 06:18:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 08:18:14 -0500 Subject: [Freeswitch-users] Freeswitch taking too long to start up In-Reply-To: References: <1B99233662E2104983E3550185D3ED7355A836@xena.internal.datapro.co.za> Message-ID: <59F86087-9012-40D0-87F7-4721E49FAA7F@freeswitch.org> You shouldn't have to use the switch anymore. That is unless you just wanna skip that check. /b On Jun 3, 2009, at 12:09 AM, Muhammad Shahzad wrote: > I had to upgrade again svn revision to use this switch, but it works. > > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/26ff7cca/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 3 06:18:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:18:50 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> References: <28083808.277891244027846976.JavaMail.root@zimbra.tng.de> <2376868.277911244027905385.JavaMail.root@zimbra.tng.de> Message-ID: <191c3a030906030618g2f6bec95lcbfe966565aefd7a@mail.gmail.com> This is called RTP Auto Adjust This occurs when the SDP of the other side sends FS the wrong media IP in the SDP If FS gets packets from some other place besides where it thinks its supposed to send packets in a window of the first 10 packets repeatedly, then it auto adjusts the destination, fixing the problem. If you look at your FreeSWITCH console when you make this call it's likely you see a message about RTP auto adjusting the IP. On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert wrote: > Hello, > > I have a question about the RTP stream. I made two calls with different > devices. One had no problems, the other call made some difficulties. I put > an extraction of my traces in attachment. > > (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) > The first one had no problem. When I called the freeSWITCH I had a > bidirectional RTP stream. I heard the announcement of the server. I could > transmit digits from my telephone to the freeSWITCH which were verified by > the machine. > > (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 > phone) > The second call had only a oneway audio stream. When I called the > freeSWITCH I heard no announcement of the server which should be actually > played. There was only a background noise. I made traces from this example. > > My suggestion is: > In packet 378 the freeSWITCH server wants to sent RTP packets to an > suspicious IP. There are over 5 packets in number. Not till then there is > the correct destination IP (see packet 386). But this could be the fact that > the freeSWITCH produces an error. > > Does anybody have an idea? > > Greetz > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/3e86bfab/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 3 06:28:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:28:02 -0500 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> Message-ID: <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> also press f8 before you take the console log to get the debugging info and paste the resulting trace in http://pastebin.freeswitch.org rather than right in the email -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/222da725/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 3 06:39:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 3 Jun 2009 08:39:21 -0500 Subject: [Freeswitch-users] Passthru mode In-Reply-To: <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> References: <302665.69883.qm@web34301.mail.mud.yahoo.com> <1FC13F0C-D57E-4735-8C42-AAF37F6F9A2A@avgs.ca> Message-ID: <191c3a030906030639wfa33c72qe265631699a00fb9@mail.gmail.com> I am pretty sure inbound-proxy-media forces late-negotation iirc. On Wed, Jun 3, 2009 at 1:33 AM, Mathieu Rene wrote: > > On 3-Jun-09, at 2:32 AM, Jim Burke wrote: > > > Fernando, > > > > Try setting 'inbound-late-negotiation' in your SIP Profile. This will > > allow the call to hit the dialplan where you can set proxy_media. > > This also assumes you have bypass_media set to false in your dialplan. > > > > Alternatively I beleive you can set "inbound-proxy-media" in the SIP > > Profile and this will do the same thing. > > But you still need late negotiation for that to work, so in both cases > you need to fix that :D > > Math > > > > > > > Regards, > > Jim > > > > On Wed, Jun 3, 2009 at 6:46 AM, FERNANDO VILLARROEL > > wrote: > >> > >> Dear, > >> > >> I can't solve my problem, i was try with: > >> > >> > >> > >> > >> and: > >> > >> in freeswitch.xml > >> > >> But receive the same log: > >> > >> http://pastebin.freeswitch.org/9204 > >> > >> Anyone help me. > >> > >> Fernando > >> > >> --- On Mon, 6/1/09, FERNANDO VILLARROEL > >> wrote: > >> > >>> From: FERNANDO VILLARROEL > >>> Subject: Re: [Freeswitch-users] Passthru mode > >>> To: freeswitch-users at lists.freeswitch.org > >>> Date: Monday, June 1, 2009, 8:10 PM > >>> > >>> Hello i was try with: > >>> > >>> > >>> >>> data="sofia/gateway/ubb/$1$2$3"/> > >>> > >>> This is the log on FS_CLI: > >>> > >>> http://pastebin.freeswitch.org/9204 > >>> > >>> Fernando > >>> > >>> --- On Mon, 6/1/09, Michael Collins > >>> wrote: > >>> > >>>> From: Michael Collins > >>>> Subject: Re: [Freeswitch-users] Passthru mode > >>>> To: freeswitch-users at lists.freeswitch.org > >>>> Date: Monday, June 1, 2009, 7:41 PM > >>>> > >>>> > >>>> On Mon, Jun 1, 2009 at 3:20 PM, > >>>> FERNANDO VILLARROEL > >>>> wrote: > >>>> > >>>> > >>>> > >>>> Hello the dial plan: > >>>> > >>>> > >>>> > >>>> >>>> data="sofia/gateway/ubb/$1$2$3"/> > >>>> > >>>> > >>>> > >>>> This i setup from Wikipbx. > >>>> What about this in the dialplan? > >>>> >>>> data="proxy_media=true"/> > >>>> Or alternatively this in the SIP profile? > >>>> > >>>> >>>> value="true"/> > >>>> > >>>> I just want to make sure you're actually telling FS > >>> to > >>>> use proxy media. If I may make a suggestion: use > >>> pastebin.freeswitch.org > >>>> and pastebin the entire extension in the dialplan as > >>> well as > >>>> a complete debug log of the call from the FS CLI. > >>> Please see > >>>> this page for some handy tips on gathering information > >>> for > >>>> troubleshooting: > >>>> > >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs > >>>> > >>>> -MC > >>>> > >>>> > >>>> > >>>> -----Inline Attachment Follows----- > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/7551d211/attachment-0002.html From asannucci at gmail.com Wed Jun 3 08:03:51 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:03:51 +0200 Subject: [Freeswitch-users] Error sofia_reg_c Message-ID: <7D3E2C7F9090451F9809F828584780B1@voztovoice> This is my problem: i have two gateway configured on FS that working fine (registered) when i start FS on the fs_cli I receive this message [ERR] sofia_reg.c:1499 sofia_reg_handle_sip_r_challenge() No Matching gateway found What means this error? Thank you. Best regards From brian at freeswitch.org Wed Jun 3 08:09:23 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:09:23 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <7D3E2C7F9090451F9809F828584780B1@voztovoice> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> Message-ID: <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> It means the far side send us a 407 and we couldn't match it to any gateway on your system to answer the challenge so we have no choice but to fail the call. /b On Jun 3, 2009, at 10:03 AM, bakko wrote: > This is my problem: > > i have two gateway configured on FS that working fine (registered) > > when i start FS on the fs_cli I receive this message > > [ERR] sofia_reg.c:1499 sofia_reg_handle_sip_r_challenge() No Matching > gateway found > > What means this error? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/5bb77b80/attachment-0002.html From matthew.lockwood at gmail.com Wed Jun 3 04:52:16 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 04:52:16 -0700 Subject: [Freeswitch-users] Softphone configuration Message-ID: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> Hi, I've just managed to get FreeSWITCH installed. I'm using the default config files which works fine with X-Lite. The problem is, I can't use any other softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is having issues with my Netgear router (it causes it to continually restart ... there are other posts about this elsewhere and the solutions aren't working.) The only other relevant thing I can think of to add to this topic is that X-Lite, on its first registration always gives me a timeout error, and then successfully registers. After a few minutes, the router reboots. Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/04970b99/attachment-0002.html From matthew.lockwood at gmail.com Wed Jun 3 06:20:53 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 06:20:53 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> Message-ID: <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> Hi, I've just managed to get FreeSWITCH installed. I'm using the default config files which works fine with X-Lite. The problem is, I can't use any other softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is having issues with my Netgear router (it causes it to continually restart ... there are other posts about this elsewhere and the solutions aren't working.) The only other relevant thing I can think of to add to this topic is that X-Lite, on its first registration always gives me a timeout error, and then successfully registers. After a few minutes, the router reboots. Matt PS. I apologize if this posts twice - I seemed to have an issue with my mail client and I don't think it sent the first time. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/198c78e7/attachment-0002.html From asannucci at gmail.com Wed Jun 3 08:16:26 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:16:26 +0200 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: Ok. But why i receive a 407 response if no call in progress or active. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/710307be/attachment-0002.html From brian at freeswitch.org Wed Jun 3 08:23:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:23:18 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <5A7C2AFB-3E3B-4212-A81B-0830A1D629A1@freeswitch.org> Ok again just guessing because you failed to provide any info but the one line in your first email... could be a bad gateway name? no clue since you guessed we only needed to see the ONE line. I would put the log on pastebin join #freeswitch on IRC and ask.. this email stuff is too slow. /b On Jun 3, 2009, at 10:16 AM, bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. > > Regards. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/54b6f223/attachment-0002.html From brian at freeswitch.org Wed Jun 3 08:18:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:18:00 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> Message-ID: <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> I'm going to have to guess that you're doing this all on the same machine? /b On Jun 3, 2009, at 8:20 AM, Matthew Lockwood wrote: > Hi, > > I've just managed to get FreeSWITCH installed. I'm using the default > config files which works fine with X-Lite. The problem is, I can't > use any other softphone other than X-Lite. i've tried YakaPhone, > Zoiper, Adore Softphone, QuteCom, and Linphone. I'd normally not say > this is a problem, but X-Lite is having issues with my Netgear > router (it causes it to continually restart ... there are other > posts about this elsewhere and the solutions aren't working.) > > The only other relevant thing I can think of to add to this topic is > that X-Lite, on its first registration always gives me a timeout > error, and then successfully registers. After a few minutes, the > router reboots. > > Matt > > PS. I apologize if this posts twice - I seemed to have an issue with > my mail client and I don't think it sent the first time. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0728eb15/attachment-0002.html From intralanman at freeswitch.org Wed Jun 3 08:23:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 11:23:58 -0400 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <4A26958E.4040503@freeswitch.org> bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. REMEMBER: ngrep is your friend. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/bb88cc34/attachment-0002.html From rdenert at tng.de Wed Jun 3 08:29:45 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 17:29:45 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <17870203.282961244042976688.JavaMail.root@zimbra.tng.de> Message-ID: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> How can I avoid the problem? It seems that RTP auto adjust generates the error. Maybe I can deactivate RTP auto adjust but I suspect that freeSWITCH doesn't find the right media IP. Do you have any other solution? Greetz ----- Urspr?ngliche Mail ----- Von: "Anthony Minessale" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 3. Juni 2009 15:18:50 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Insufficient RTP stream This is called RTP Auto Adjust This occurs when the SDP of the other side sends FS the wrong media IP in the SDP If FS gets packets from some other place besides where it thinks its supposed to send packets in a window of the first 10 packets repeatedly, then it auto adjusts the destination, fixing the problem. If you look at your FreeSWITCH console when you make this call it's likely you see a message about RTP auto adjusting the IP. On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have a question about the RTP stream. I made two calls with different devices. One had no problems, the other call made some difficulties. I put an extraction of my traces in attachment. (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 phone) The first one had no problem. When I called the freeSWITCH I had a bidirectional RTP stream. I heard the announcement of the server. I could transmit digits from my telephone to the freeSWITCH which were verified by the machine. (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset 5020 phone) The second call had only a oneway audio stream. When I called the freeSWITCH I heard no announcement of the server which should be actually played. There was only a background noise. I made traces from this example. My suggestion is: In packet 378 the freeSWITCH server wants to sent RTP packets to an suspicious IP. There are over 5 packets in number. Not till then there is the correct destination IP (see packet 386). But this could be the fact that the freeSWITCH produces an error. Does anybody have an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From matthew.lockwood at gmail.com Wed Jun 3 08:34:06 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 08:34:06 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> Message-ID: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> No, FS is installed on a VPS. I'm just connecting through my cable connection. M On Wed, Jun 3, 2009 at 8:18 AM, Brian West wrote: > I'm going to have to guess that you're doing this all on the same machine? > /b > > On Jun 3, 2009, at 8:20 AM, Matthew Lockwood wrote: > > Hi, > > I've just managed to get FreeSWITCH installed. I'm using the default config > files which works fine with X-Lite. The problem is, I can't use any other > softphone other than X-Lite. i've tried YakaPhone, Zoiper, Adore Softphone, > QuteCom, and Linphone. I'd normally not say this is a problem, but X-Lite is > having issues with my Netgear router (it causes it to continually restart > ... there are other posts about this elsewhere and the solutions aren't > working.) > > The only other relevant thing I can think of to add to this topic is that > X-Lite, on its first registration always gives me a timeout error, and then > successfully registers. After a few minutes, the router reboots. > > Matt > > PS. I apologize if this posts twice - I seemed to have an issue with my > mail client and I don't think it sent the first time. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/53f52dfa/attachment-0002.html From rdenert at tng.de Wed Jun 3 08:35:17 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 3 Jun 2009 17:35:17 +0200 (CEST) Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <31324436.283071244043300883.JavaMail.root@zimbra.tng.de> Message-ID: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> Is there a possibility to activate all DTMF detection modes (in-band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same extension of the dialplan? Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 21:29:19 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. Actually, that's very interesting. The start_dtmf app looks for in-band DTMFs, so if you have to have that in order to hear digits from the other end then the other end most definitely isn't doing RFC2833. For the sake of testing, try sending in-band and see how the other end reacts. Might want to check this out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate My guess is that the equipment along the way is futzing with things. FreeSWITCH does break easily but it does find bugs in other VoIP systems that it talks to... :) -MC The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Urspr?ngliche Mail ----- Von: "Michael Collins" < msc at freeswitch.org > An: freeswitch-users at lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert at tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From intralanman at freeswitch.org Wed Jun 3 08:36:42 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 11:36:42 -0400 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: <4A26988A.1040308@freeswitch.org> Matthew Lockwood wrote: > No, FS is installed on a VPS. I'm just connecting through my cable > connection. > SPI or SIP ALG on the router? -Ray From brian at freeswitch.org Wed Jun 3 08:38:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:38:45 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: I'm guessing NAT problem, Not sure those other phones do STUN and traverse nat properly. /b On Jun 3, 2009, at 10:34 AM, Matthew Lockwood wrote: > No, FS is installed on a VPS. I'm just connecting through my cable > connection. > > M Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/5f1b436e/attachment-0002.html From brian at freeswitch.org Wed Jun 3 08:39:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:39:42 -0500 Subject: [Freeswitch-users] Prolems with DTMF In-Reply-To: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> References: <13160662.283091244043317389.JavaMail.root@zimbra.tng.de> Message-ID: NO. You should only have one at a time. Its impossible to handle the scenario where you receive say on 2833 and then end up sending an info packet, 2833 packet and the inband of the DTMF... triple DTMF. :) /b On Jun 3, 2009, at 10:35 AM, Rudolf Denert wrote: > Is there a possibility to activate all DTMF detection modes (in- > band, SIP INFO & RFC 2388) in the same dialplan or maybe in the same > extension of the dialplan? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/8e5be88c/attachment-0002.html From dujinfang at gmail.com Wed Jun 3 08:41:46 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 23:41:46 +0800 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> Message-ID: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> I had this problem when I gateway to an asterisk box. Each time I call to asterisk through that gateway got a 407 and fail. Never figured out why but guess it's non-proper configuration of Asterisk. On Jun 3, 2009, at 11:16 PM, bakko wrote: > Ok. > > But why i receive a 407 response if no call in progress or active. > > Regards. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/d17d9a1d/attachment-0002.html From matthew.lockwood at gmail.com Wed Jun 3 08:42:30 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 08:42:30 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <4A26988A.1040308@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> Message-ID: <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> SPI disabled. There's no obvious option to disable SIP ALG, but NAT filtering was changed from secure to open. Problem still persists. On Wed, Jun 3, 2009 at 8:36 AM, Raymond Chandler wrote: > Matthew Lockwood wrote: > > No, FS is installed on a VPS. I'm just connecting through my cable > > connection. > > > SPI or SIP ALG on the router? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/eda81d32/attachment-0002.html From dujinfang at gmail.com Wed Jun 3 08:43:08 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 3 Jun 2009 23:43:08 +0800 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> References: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> Message-ID: <413F2852-3088-4035-BA26-B84C5BFE51EA@gmail.com> have you tried this? http://wiki.freeswitch.org/wiki/Channel_Variables#disable_rtp_auto_adjust On Jun 3, 2009, at 11:29 PM, Rudolf Denert wrote: > How can I avoid the problem? It seems that RTP auto adjust generates > the error. Maybe I can deactivate RTP auto adjust but I suspect that > freeSWITCH doesn't find the right media IP. > > Do you have any other solution? > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Anthony Minessale" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Mittwoch, 3. Juni 2009 15:18:50 GMT +01:00 Amsterdam/ > Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Insufficient RTP stream > > > This is called RTP Auto Adjust > This occurs when the SDP of the other side sends FS the wrong media > IP in the SDP > If FS gets packets from some other place besides where it thinks its > supposed to send packets in a window of the first 10 packets > repeatedly, > then it auto adjusts the destination, fixing the problem. If you > look at your FreeSWITCH console when you make this call it's likely > you see a message about RTP auto adjusting the IP. > > > > On Wed, Jun 3, 2009 at 6:18 AM, Rudolf Denert < rdenert at tng.de > > wrote: > > > Hello, > > I have a question about the RTP stream. I made two calls with > different devices. One had no problems, the other call made some > difficulties. I put an extraction of my traces in attachment. > > (MGCP device -> Thomson Speedtouch 780WL with a Siemens Euroset 5020 > phone) > The first one had no problem. When I called the freeSWITCH I had a > bidirectional RTP stream. I heard the announcement of the server. I > could transmit digits from my telephone to the freeSWITCH which were > verified by the machine. > > (VoATM Device -> Allied Data Copperjet 1616 with a Siemens Euroset > 5020 phone) > The second call had only a oneway audio stream. When I called the > freeSWITCH I heard no announcement of the server which should be > actually played. There was only a background noise. I made traces > from this example. > > My suggestion is: > In packet 378 the freeSWITCH server wants to sent RTP packets to an > suspicious IP. There are over 5 packets in number. Not till then > there is the correct destination IP (see packet 386). But this could > be the fact that the freeSWITCH produces an error. > > Does anybody have an idea? > > Greetz > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. > Any > unauthorized copying, disclosure or distribution of the material in > this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. > Any > unauthorized copying, disclosure or distribution of the material in > this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From asannucci at gmail.com Wed Jun 3 08:41:54 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 17:41:54 +0200 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com><415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com><7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> Message-ID: Have you open the necesary ports in the VPS firewall? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/57d8716b/attachment-0002.html From brian at freeswitch.org Wed Jun 3 08:45:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:45:44 -0500 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice> <96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> Message-ID: <1977D7D9-E777-40DC-A24D-AB4894471E6F@freeswitch.org> Its because its challenging you and you can't answer the challenge... its saying HEY YOU give me a user/pass ... and you can't answer that so it fails. /b On Jun 3, 2009, at 10:41 AM, dujinfang wrote: > I had this problem when I gateway to an asterisk box. Each time I > call to asterisk through that gateway got a 407 and fail. Never > figured out why but guess it's non-proper configuration of Asterisk. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/1626d3d0/attachment-0002.html From brian at freeswitch.org Wed Jun 3 08:46:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:46:14 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> References: <22414518.282981244042985393.JavaMail.root@zimbra.tng.de> Message-ID: <17D2F89B-F6ED-4DD7-9C1F-FEDBD03F7D67@freeswitch.org> Without a sip trace its hard to tell. /b On Jun 3, 2009, at 10:29 AM, Rudolf Denert wrote: > How can I avoid the problem? It seems that RTP auto adjust generates > the error. Maybe I can deactivate RTP auto adjust but I suspect that > freeSWITCH doesn't find the right media IP. > > Do you have any other solution? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/ac802066/attachment-0002.html From brian at freeswitch.org Wed Jun 3 08:47:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 10:47:30 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> Message-ID: Thats not the problem.. Phone registers to freeswitch has 192.168.0.100 in the sip packet... FS sends a challenge back to 192.168.0.100 which obviously will FAIL cuz that network isn't anywhere near the VPS box... Go to the FreeSWITCH box and type "sofia profile internal siptrace on" and I'll suspect you see this exact behavior. /b On Jun 3, 2009, at 10:42 AM, Matthew Lockwood wrote: > SPI disabled. There's no obvious option to disable SIP ALG, but NAT > filtering was changed from secure to open. Problem still persists. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/95effb84/attachment-0002.html From asannucci at gmail.com Wed Jun 3 09:24:35 2009 From: asannucci at gmail.com (bakko) Date: Wed, 3 Jun 2009 18:24:35 +0200 Subject: [Freeswitch-users] Error sofia_reg_c In-Reply-To: <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> References: <7D3E2C7F9090451F9809F828584780B1@voztovoice><96AC6D7E-2C28-4E3B-95BF-681F7BB35B48@freeswitch.org> <8EA44826-469F-41DE-868C-82612DE1342C@gmail.com> Message-ID: This is not my problem. I have conected the asterisk to FS and i can call from FS to Asterisk without problem. I do some tests to call a international number using a sip provider that is registered on the asterisk pbx and work. Actualy i can: call from any FS extension any Asterisk extension call from any asterisk extension any fs extension I tried to disable the asterisk gateway configuration but not resolve the issue. Maybe the problem is with the other gateway. I have to investigate :) Sorry for my very bad english :) Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/b18dd2ce/attachment-0002.html From fdelawarde at wirelessmundi.com Wed Jun 3 09:30:40 2009 From: fdelawarde at wirelessmundi.com (Francois Delawarde) Date: Wed, 03 Jun 2009 18:30:40 +0200 Subject: [Freeswitch-users] video transcoding Message-ID: <1244046640.28699.63.camel@localhost.localdomain> Hello, I'm interested in being able to do video transcoding mainly for bridging 3G mobile and sip networks, and maybe later on some conferencing with FS. Are video codecs planned to be added to FS even in a far future? Are there copyright/patent problems with common video codecs (H.263 / H.264) or with libraries (ffmpeg) that would prevent any of that from happening? Meanwhile, would it be feasible to do some video transcoding using external software (vlc?) with socket connections from-to FS? Thanks, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/c4d1d377/attachment-0002.html From matthew.lockwood at gmail.com Wed Jun 3 10:00:32 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:00:32 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> Message-ID: <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> When I type that verbatim, I get the following error: -ERR Unknown command! M On Wed, Jun 3, 2009 at 8:47 AM, Brian West wrote: > Thats not the problem.. > Phone registers to freeswitch has 192.168.0.100 in the sip packet... FS > sends a challenge back to 192.168.0.100 which obviously will FAIL cuz that > network isn't anywhere near the VPS box... > > > Go to the FreeSWITCH box and type "sofia profile internal siptrace on" and > I'll suspect you see this exact behavior. > > /b > > > > On Jun 3, 2009, at 10:42 AM, Matthew Lockwood wrote: > > SPI disabled. There's no obvious option to disable SIP ALG, but NAT > filtering was changed from secure to open. Problem still persists. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0ec49beb/attachment-0002.html From brian at freeswitch.org Wed Jun 3 10:04:33 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 12:04:33 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> Message-ID: <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> Update to SVN trunk. /b On Jun 3, 2009, at 12:00 PM, Matthew Lockwood wrote: > When I type that verbatim, I get the following error: > > -ERR Unknown command! > > M Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/4dd8ad84/attachment-0002.html From matthew.lockwood at gmail.com Wed Jun 3 10:11:42 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:11:42 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> Message-ID: <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> Is that stable enough to use in production? On Wed, Jun 3, 2009 at 10:04 AM, Brian West wrote: > Update to SVN trunk. > /b > > On Jun 3, 2009, at 12:00 PM, Matthew Lockwood wrote: > > When I type that verbatim, I get the following error: > > -ERR Unknown command! > > M > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/9b57c7b9/attachment-0002.html From intralanman at freeswitch.org Wed Jun 3 10:16:05 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 03 Jun 2009 13:16:05 -0400 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> Message-ID: <4A26AFD5.1080903@freeswitch.org> Matthew Lockwood wrote: > Is that stable enough to use in production? it's more stable than "not working" -Ray From brian at freeswitch.org Wed Jun 3 10:34:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 3 Jun 2009 12:34:12 -0500 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <4A26AFD5.1080903@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030620j31099ba2nb40429fef12d6d21@mail.gmail.com> <7F564399-ED6E-4AC5-A462-F7D488E1B098@freeswitch.org> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> <4A26AFD5.1080903@freeswitch.org> Message-ID: <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> It usually is... if you pop on IRC people can tell you one way or another. Issues never hang around for very long! /b On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote: > Matthew Lockwood wrote: >> Is that stable enough to use in production? > it's more stable than "not working" > -Ray Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/c599f0dc/attachment-0002.html From matthew.lockwood at gmail.com Wed Jun 3 10:39:02 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Wed, 3 Jun 2009 10:39:02 -0700 Subject: [Freeswitch-users] Softphone configuration In-Reply-To: <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> References: <415541b10906030452s19c629fi1d65bc157ed19ba6@mail.gmail.com> <415541b10906030834w585abf12m29cdefa6eba55f54@mail.gmail.com> <4A26988A.1040308@freeswitch.org> <415541b10906030842o3f2e14cfq32f5c06a112c7d24@mail.gmail.com> <415541b10906031000o11ee5634gd06c312747c18f92@mail.gmail.com> <6FD15EE1-FDE7-4037-9F05-5486DCAACE0B@freeswitch.org> <415541b10906031011x50a7eac6q56ea7a154a763271@mail.gmail.com> <4A26AFD5.1080903@freeswitch.org> <31812B24-AE9E-4972-9BF6-DFB7D2E0EEF5@freeswitch.org> Message-ID: <415541b10906031039s49832117m8972e557f6875760@mail.gmail.com> It's installing now. I'll get back with the results shortly. On Wed, Jun 3, 2009 at 10:34 AM, Brian West wrote: > It usually is... if you pop on IRC people can tell you one way or another. > Issues never hang around for very long! > /b > > On Jun 3, 2009, at 12:16 PM, Raymond Chandler wrote: > > Matthew Lockwood wrote: > > Is that stable enough to use in production? > > it's more stable than "not working" > -Ray > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/0d3cea27/attachment-0002.html From testeador01 at gmail.com Wed Jun 3 10:39:25 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 3 Jun 2009 12:39:25 -0500 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: <20090603063909.GA19487@jdc.jasonjgw.net> References: <20090603063909.GA19487@jdc.jasonjgw.net> Message-ID: Hi Woody :) You cannot use the variable on another extension, however you could just merge both extensions' conditions. then your only problem would be that you're not exporting the value, after set, you gotta export, look at this example (a little extract from dialplan/default.xml): ... 2009/6/3 Jason White > Woody Dickson wrote: > > I am getting a strange problem in my dialplan. > > > > After doing "SET", I want to use it in the next condition field. But > then > > the value is not being set properly. > > When parsing the dial plan, FreeSWITCH tests all of the conditions, then > builds a linked list of actions to execute. Once this is done, the actions > are > executed, in order. > > This is why you can't simply set a variable in one extension and test it in > the condition of a later extension. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/64af6b64/attachment-0002.html From mrene_lists at avgs.ca Wed Jun 3 10:43:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 13:43:18 -0400 Subject: [Freeswitch-users] Set problem in dialplan In-Reply-To: References: <20090603063909.GA19487@jdc.jasonjgw.net> Message-ID: <2B642D7C-2779-4BF4-B3AF-017CDD4CE3CE@avgs.ca> Export transfers the variable to the B-leg whenever the channel is bridged, it doesnt affect how the dialplan work, conditions are still checked before executing anything. Math On 3-Jun-09, at 1:39 PM, Milena wrote: > Hi Woody :) > You cannot use the variable on another extension, however you could > just merge both extensions' conditions. > > then your only problem would be that you're not exporting the value, > after set, you gotta export, look at this example (a little extract > from dialplan/default.xml): > > > > > > > > > > > > ... > > > > > > > 2009/6/3 Jason White > Woody Dickson wrote: > > I am getting a strange problem in my dialplan. > > > > After doing "SET", I want to use it in the next condition field. > But then > > the value is not being set properly. > > When parsing the dial plan, FreeSWITCH tests all of the conditions, > then > builds a linked list of actions to execute. Once this is done, the > actions are > executed, in order. > > This is why you can't simply set a variable in one extension and > test it in > the condition of a later extension. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/49c442ef/attachment-0002.html From larclap at yahoo.com Wed Jun 3 11:04:22 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 3 Jun 2009 11:04:22 -0700 Subject: [Freeswitch-users] Dialplan appliction db? Message-ID: <053901c9e475$b91e3140$2b5a93c0$@com> Anyone point me to the wiki which describes the "db" application and its arguments? The following is a snippet from a dialplan. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/dc902e77/attachment-0002.html From mrene_lists at avgs.ca Wed Jun 3 11:06:03 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 3 Jun 2009 14:06:03 -0400 Subject: [Freeswitch-users] Dialplan appliction db? In-Reply-To: <053901c9e475$b91e3140$2b5a93c0$@com> References: <053901c9e475$b91e3140$2b5a93c0$@com> Message-ID: <327DDDBA-A6B8-4243-AAFE-260B466870E7@avgs.ca> freeswitch at Maths-Mac.local> show api db shAPI CALL [show(api db)] output: name,description,syntax,key db,db get/set,[insert|delete|select]///,mod_limit 1 total. freeswitch at Maths-Mac.local> show api hash API CALL [show(api hash)] output: name,description,syntax,key hash,hash get/set,[insert|delete|select]///,mod_limit 1 total. freeswitch at Maths-Mac.local> On 3-Jun-09, at 2:04 PM, Lars Zeb wrote: > Anyone point me to the wiki which describes the ?db? application and > its arguments? The following is a snippet from a dialplan. > > > > > > > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090603/cd42a759/attachment-0002.html From larclap at yahoo.com Wed Jun 3 11:26:57 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 3 Jun 2009 11:26:57 -0700 Subject: [Freeswitch-users] Error reloadxml at console Message-ID: <055c01c9e478$e1163ff0$a342bfd0$@com> When I type reloadxml at fs console, I get the following message: freeswitch at fs> reloadxml API CALL [reloadxml()] output: +OK [[error near line 3182]: unclosed ). You warned me about this in an earlier email. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 03, 2009 11:52 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error reloadxml at console On Wed, Jun 3, 2009 at 11:26 AM, Lars Zeb wrote: When I type reloadxml at fs console, I get the following message: freeswitch at fs> reloadxml API CALL [reloadxml()] output: +OK [[error near line 3182]: unclosed when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/7b38836d/attachment-0002.html From mike at jerris.com Thu Jun 4 01:59:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 04:59:34 -0400 Subject: [Freeswitch-users] Error causing freeswitch to crash In-Reply-To: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> References: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please attempt to reproduce this issue with trunk with crash protection disabled, and if you are able please file a jira with a backtrace of the crash Mike On Jun 4, 2009, at 4:23 AM, Andy Ayers wrote: > Hi, > > Every few days I'm getting this error which is causing Freeswitch to > crash. Can anyone tell me what may be causing this or how to prevent > it? > > 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 > handle_fatality() Caught signal 11 for unmapped thread! > > Many thanks > Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/6c2535aa/attachment-0002.html From mike at jerris.com Thu Jun 4 02:02:52 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 05:02:52 -0400 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604082921.GA838@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> Message-ID: Can you please re-test with current svn trunk. we added some new nat busting code yesterday that may assist with this. You will need to specify the new param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 ) Mike On Jun 4, 2009, at 4:29 AM, Jason White wrote: > I hate NAT with a passion that strengthens by the day! > > I'm trying to interact with my ISP, which is a SIP provider. My FS > system is > behind a Cisco router (IOS 12.4(6)t). The provider recommends > turning off SIP > handling in the router's NAT configuration due to bugs in this > version of IOS. > I have found that I need to do this, otherwise incoming calls are > never > received, even though outgoing calls work without this change in > place. > > In the router: > no ip nat service sip udp port 5060 > ip nat inside source static tcp 192.168.0.2 5080 interface Dialer1 > 5080 > ip nat inside source static udp 192.168.0.2 5080 interface Dialer1 > 5080 > > With this change, incoming calls to FS are fine, but outgoing calls > are not > (see the SIP trace below). > > However, if I register to the same provider from an internal Snom > 320 phone > and make a call, it works. > > FreeSWITCH has sophisticated NAT handling features, as does the > other side of > this connection, and somehow they aren't working together. (I don't > know how > the other end is set up, but they claim to have complex NAT handling > logic). > > Registration is successful, by the way. > > The 118.208.xxx.xxx address is mine, dynamically allocated by the > ISP. The > xxxxxxxxxx at sip.internode.on.net address is my user name/address at > the service > provider. > > freeswitch at default> sofia profile external siptrace on > Enabled sip debugging on external > freeswitch at default> send 948 bytes to udp/[203.2.134.1]:5060 at > 07:14:51.882839: > > ------------------------------------------------------------------------ > REGISTER sip:sip.internode.on.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK4m9SgBce1DXpc > Max-Forwards: 70 > From: > ;tag=jy38KK47549va > To: > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931044 REGISTER > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj48x4rTpg7mg2BW", > cnonce="8SrBsct3EiylvAAaS8o90w", algorithm=MD5, uri="sip:sip.internode.on.net;transport=udp > ", response="85ec38442fbd26153bacd99c659bd037", qop=auth, nc=0000001c > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[203.2.134.1]:5060 at 07:14:51.967127: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK4m9SgBce1DXpc;rport=5080 > From: > ;tag=jy38KK47549va > To: > < > sip:xxxxxxxxxx > @sip.internode.on.net;transport=udp>;tag=aprqcauh8h3-4d3bh0p08vt9a > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931044 REGISTER > Contact: >;expires=60 > > > ------------------------------------------------------------------------ > 2009-06-04 17:14:53 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/internal/ > 1000 at 192.168.0.2:5070 [6653f750-50d7-11de-b1c2-25f4151d7bef] > 2009-06-04 17:14:53 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing Extension 1000->90871271202 in context default > 2009-06-04 17:14:53 [NOTICE] mod_sofia.c:1430 > sofia_receive_message() Ring-Ready sofia/internal/ > 1000 at 192.168.0.2:5070! > 2009-06-04 17:14:53 [NOTICE] mod_dptools.c:415 ring_ready_function() > Ring Ready sofia/internal/1000 at 192.168.0.2:5070! > 2009-06-04 17:14:53 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/external/0871271202 > [66551aea-50d7-11de-b1c2-25f4151d7bef] > send 1242 bytes to udp/[203.2.134.1]:5060 at 07:14:53.275254: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK5X2jj6vHypK9Q > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > recv 326 bytes from udp/[203.2.134.1]:5060 at 07:14:53.362415: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK5X2jj6vHypK9Q;rport=5080 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > > > ------------------------------------------------------------------------ > recv 487 bytes from udp/[203.2.134.1]:5060 at 07:14:53.371970: > > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK5X2jj6vHypK9Q;rport=5080 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: 0871271202 at sip.internode.on.net>;tag=1512759423-1244099693344 > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 INVITE > WWW-Authenticate: DIGEST > qop > = > "auth > ",nonce > ="BroadWorksXfvj4u334T7pbfc4BW",algorithm=MD5,realm="BroadWorks" > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 388 bytes to udp/[203.2.134.1]:5060 at 07:14:53.372231: > > ------------------------------------------------------------------------ > ACK sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK5X2jj6vHypK9Q > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: 0871271202 at sip.internode.on.net>;tag=1512759423-1244099693344 > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931510 ACK > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:53.372622: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:53.873867: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:54.873857: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:14:56.873871: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:00.873875: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:08.879056: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:24.881893: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:25.375565: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:25.877859: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:26.877862: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:28.878002: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:32.881880: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > send 948 bytes to udp/[203.2.134.1]:5060 at 07:15:32.898046: > > ------------------------------------------------------------------------ > REGISTER sip:sip.internode.on.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK7FN4Nvyrr8ZeF > Max-Forwards: 70 > From: > ;tag=jy38KK47549va > To: > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931045 REGISTER > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj48x4rTpg7mg2BW", > cnonce="8SrBsct3EiylvAAaS8o90w", algorithm=MD5, uri="sip:sip.internode.on.net;transport=udp > ", response="2e9f91b4c82114b7c52a6c162633fbf2", qop=auth, nc=0000001d > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 453 bytes from udp/[203.2.134.1]:5060 at 07:15:33.017609: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.0.2 > :5080;received=118.208.xxx.xxx;branch=z9hG4bK7FN4Nvyrr8ZeF;rport=5080 > From: > ;tag=jy38KK47549va > To: > < > sip:xxxxxxxxxx > @sip.internode.on.net;transport=udp>;tag=aprqcauh8h3-4d3bh0p08vtba > Call-ID: 19ad6c76-50d5-11de-b1c2-25f4151d7bef > CSeq: 115931045 REGISTER > Contact: >;expires=60 > > > ------------------------------------------------------------------------ > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:40.882383: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > 2009-06-04 17:15:53 [NOTICE] switch_ivr_originate.c:1957 > switch_ivr_originate() Hangup sofia/external/0871271202 > [CS_CONSUME_MEDIA] [NO_ANSWER] > 2009-06-04 17:15:53 [INFO] mod_dptools.c:2091 > audio_bridge_function() Originate Failed. Cause: NO_ANSWER > 2009-06-04 17:15:53 [NOTICE] switch_core_state_machine.c:179 > switch_core_standard_on_execute() Hangup sofia/internal/1000 at 192.168.0.2 > :5070 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/internal/1000 at 192.168.0.2 > :5070) Ended > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/internal/1000 at 192.168.0.2 > :5070 [CS_DESTROY] > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 8 (sofia/external/0871271202) > Ended > 2009-06-04 17:15:53 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/external/0871271202 > [CS_DESTROY] > send 1518 bytes to udp/[203.2.134.1]:5060 at 07:15:56.882385: > > ------------------------------------------------------------------------ > INVITE sip:0871271202 at sip.internode.on.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.2:5080;rport;branch=z9hG4bK66UBm1DNUZ9UK > Max-Forwards: 69 > From: "Jason White" >;tag=ZKcBBKZN0rBgH > To: > Call-ID: 3db38440-cb7a-122c-bca5-001a4bca3dd3 > CSeq: 115931511 INVITE > Contact: > > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Authorization: Digest username="xxxxxxxxxx", realm="BroadWorks", > nonce="BroadWorksXfvj4u334T7pbfc4BW", > cnonce="PcJk8st6EiylvAAaS8o90w", algorithm=MD5, uri="sip:0871271202 at sip.internode.on.net > ", response="d93d006a1892276d752d984ed9ab82bf", qop=auth, nc=00000001 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 360 > P-Key-Flags: keys="3" > Remote-Party-ID: "Jason White" > < > sip:xxxxxxxxxx > @sip.internode.on.net>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 6983343116146055149 6602851124936663084 IN IP4 > 192.168.0.2 > s=FreeSWITCH > c=IN IP4 192.168.0.2 > t=0 0 > m=audio 25368 RTP/AVP 9 114 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:114 CELT/48000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > > freeswitch at default> /exit > > jason at jdc:~$ exit > > Script done on Thu 04 Jun 2009 17:16:06 EST > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Jun 4 02:05:43 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 4 Jun 2009 05:05:43 -0400 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <200906041628066407689@gmail.com> References: <200906041628066407689@gmail.com> Message-ID: <84AF0BA9-7027-4D51-B5FE-9672F67D71A2@jerris.com> Are you having this issue on your analog or pri lines? what does your openzap.conf look like? Mike On Jun 4, 2009, at 4:28 AM, god.nirvana wrote: > hi all > i am new to freeswitch. > there are some busy tone detect issues,i hope someone could help me. > i installed freeswitch from trunk,openzap,zaptel.... > but i found some busy tone isuues > > my tones.conf: > [us] > generate-dial => v=-7;%(1000,0,350,440) > detect-dial => 350,440 > generate-ring => v=-7;%(2000,4000,440,480) > detect-ring => 440,480 > generate-busy => v=-7;%(500,500,450,340) > detect-busy =>450,340 > generate-attn => v=0;%(100,100,1400,2060,2450,2600) > detect-attn => 1400,2060,2450,2600 > generate-callwaiting-sas => v=0;%(300,0,440) > detect-callwaiting-sas => 440 > generate-callwaiting-cas => v=0;%(80,0,2750,2130) > detect-callwaiting-cas => 2750,2130 > detect-fail1 => 913.8 > detect-fail2 => 1370.6 > detect-fail3 => 776.7 > > > openzap.conf.xml : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > when i call the pstn phone from a ip phone,if the pstn call hangup > first,the ip phone will hear the busy tone,but the system does not > handle the busytone ,the channel does not erase. so i have to add > in the dialplan.and it works.the channel > erased. > > but in the conference case,pstn phone call in,hangup. all > participants hear the tone,"do ~,do~~".freeswitch doest handle it. > so i change the conference dialplan. > > > > > > > > > > restart freeswitch,try again,freeswitch not handle the hangup tone > still,all participants hear the tone. > how to solve it?could some one help me ??? > thx! > BR > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/72bef1c1/attachment-0002.html From dujinfang at gmail.com Thu Jun 4 02:13:01 2009 From: dujinfang at gmail.com (seven) Date: Thu, 4 Jun 2009 17:13:01 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: Message-ID: the default config allows 1002 press *1 and 1003 to do blind transfer, also you may interest the att_xfer, see dp_tools on wiki. On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: > If i don't want to use softphone function to transfer the call ,how > to do it?? > > 2009/6/4 Brian West > Depends.. Press the transfer key on your phone is how I would do > it.. what kind of phone do you have? > > /b > > On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: > >> When User(1001) calling with User(1002) , >> >> how to transfer User(1002) to User(1003)?? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/772a0d3d/attachment-0002.html From god.nirvana at gmail.com Thu Jun 4 02:15:12 2009 From: god.nirvana at gmail.com (god.nirvana) Date: Thu, 4 Jun 2009 17:15:12 +0800 Subject: [Freeswitch-users] busy tone detect issue References: <200906041628066407689@gmail.com>, <84AF0BA9-7027-4D51-B5FE-9672F67D71A2@jerris.com> Message-ID: <200906041715100317684@gmail.com> hi,thanks for your reply. my openzap.conf like this: [span zt FXO1] name => OpenZAP-FXO1 number => 1 fxo-channel => 1 [span zt FXO2] name => OpenZAP-FXO2 number => 2 fxo-channel => 2 [span zt FXO2] name => OpenZAP-FXO2 number => 3 fxo-channel => 3 [span zt FXO2] name => OpenZAP-FXO2 number => 3 fxo-channel => 4 i have a 4 fxo ports TDM400. 2009-06-04 god.nirvana ???? Michael Jerris ????? 2009-06-04 17:06:05 ???? freeswitch-users ??? ??? Re: [Freeswitch-users] busy tone detect issue Are you having this issue on your analog or pri lines? what does your openzap.conf look like? Mike On Jun 4, 2009, at 4:28 AM, god.nirvana wrote: hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 openzap.conf.xml : when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b0ad3d1f/attachment-0002.html From brad.tuan at gmail.com Thu Jun 4 02:27:10 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Thu, 4 Jun 2009 17:27:10 +0800 Subject: [Freeswitch-users] Problem about displayname of a routing call In-Reply-To: <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> References: <87f2f3b90906021102h30db87c9qd113f6bda633992e@mail.gmail.com> <191c3a030906030628p485df9d9je6b9ebd5e6a8406@mail.gmail.com> Message-ID: I know why the display name is wrong.......... in conf\directory\97730002.xml I forgot this setting........... but if I don't set cidr here ,the call from 163.28.32.51 can't come into my FS. How to make some setting for this?? 2009/6/3 Anthony Minessale > also press f8 before you take the console log to get the debugging info > and paste the resulting trace in http://pastebin.freeswitch.org rather > than right in the email > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/5c0f8fe0/attachment-0002.html From brad.tuan at gmail.com Thu Jun 4 02:36:02 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Thu, 4 Jun 2009 17:36:02 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: Message-ID: I mean when 1001 and 1002 are talking to each other , then 1001 want to transfer 1002 to 1003. 2009/6/4 seven > the default config allows 1002 press *1 and 1003 to do blind transfer, also > you may interest the att_xfer, see dp_tools on wiki. > > > On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: > > If i don't want to use softphone function to transfer the call ,how to do > it?? > > 2009/6/4 Brian West > >> Depends.. Press the transfer key on your phone is how I would do it.. what >> kind of phone do you have? >> /b >> >> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >> >> When User(1001) calling with User(1002) , >> >> how to transfer User(1002) to User(1003)?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/12f07146/attachment-0002.html From jason at jasonjgw.net Thu Jun 4 02:41:52 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 4 Jun 2009 19:41:52 +1000 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: References: <20090604082921.GA838@jdc.jasonjgw.net> Message-ID: <20090604094152.GA15717@jdc.jasonjgw.net> Michael Jerris wrote: > Can you please re-test with current svn trunk. we added some new nat > busting code yesterday that may assist with this. You will need to > specify the new > param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 > ) If I apply this to the external profile (using it on the internal profile has no effect), the Via headers now receive the external IP address, i.e., the publicly routable address provided by the ISP. However, the session description still has the private address in it, which causes the remote end to issue the following: SIP/2.0 488 Invalid Session Description Warning: 301 203.2.134.1 "invalid transport IP address" I assume (but may be wrong - full traces can be provided if necessary) that the problem is in the SDP that we're sending out: o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 Can this be fixed up? Thanks. Jason. From zhaoxxqq at 163.com Thu Jun 4 02:44:34 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Thu, 4 Jun 2009 17:44:34 +0800 Subject: [Freeswitch-users] (no subject) Message-ID: <200906041744323372036@163.com> zhaoxxqq pisze: > HI, > I use event socket to send command to FS conference. > I send " conference testconf play /root/test.wav" in console. It worked ok. > I send "api conference testconf play /record/test.wav" by event socket. > and the response is"Disconneted, Good bye.See you at ClueCon..". > I changed the wav file to www root. the same problem. can you help me? > 2009-06-01 Do you use 'auth ClueCon' before sending 'api' command? Szymon I have not use 'auth Cluecon' before sending api command. I send other api have no problem.only play wav have problems 2009-06-04 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/05377f5b/attachment-0002.html From jason at jasonjgw.net Thu Jun 4 03:39:58 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 4 Jun 2009 20:39:58 +1000 Subject: [Freeswitch-users] api conference play command In-Reply-To: <200906041744323372036@163.com> References: <200906041744323372036@163.com> Message-ID: <20090604103958.GA25407@jdc.jasonjgw.net> zhaoxxqq wrote: > I have not use 'auth Cluecon' before sending api command. > I send other api have no problem.only play wav have problems Try it from a telnet session. Start with auth ClueCon, then issue the API command as shown in my example. Unless you do something wrong, it will work. From dujinfang at gmail.com Thu Jun 4 03:51:21 2009 From: dujinfang at gmail.com (dujinfang) Date: Thu, 4 Jun 2009 18:51:21 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: Message-ID: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> yes. Did you ever tried that? On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: > I mean when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003. > 2009/6/4 seven > the default config allows 1002 press *1 and 1003 to do blind > transfer, also you may interest the att_xfer, see dp_tools on wiki. > > > On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >> If i don't want to use softphone function to transfer the call ,how >> to do it?? >> >> 2009/6/4 Brian West >> Depends.. Press the transfer key on your phone is how I would do >> it.. what kind of phone do you have? >> >> /b >> >> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >> >>> When User(1001) calling with User(1002) , >>> >>> how to transfer User(1002) to User(1003)?? >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3cf2787f/attachment-0002.html From yivzhenko at mksat.net Thu Jun 4 04:02:42 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 4 Jun 2009 14:02:42 +0300 Subject: [Freeswitch-users] mod_lcr caller id Message-ID: <200906041402.43704.yivzhenko@mksat.net> Hello. I have another problem with setting caller id with cid field. command lcr has parameter [caller_id]. and if i call lcr from the commandline it works fine, but if i call lcr from dialplan it ignores this parameter and use original caller id from caller. my cid field = /^(.*)$/999$1/ If i call lcr from the commandline lcr 444555 default 11111 lcr return origination_caller_id_number=99911111 but if i call lcr from dialplan lcr return origination_caller_id_number=9991002 1002 is a original caller id number There is a way to use this parameter from dialplan? Yuriy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/76765413/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 05:43:24 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 22:43:24 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. Message-ID: Hi All, Looking for some debugging tips and comments on what might be causing the media port in the 200OK ( Answer message) to be set to 0 by freeswitch. Essentially it looks like data might be getting trampled somehow. Portion of 200OK going into Freeswitch m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 NSE/8000..a= fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:20..a=sendrecv.. Portion of 200OK coming out of Freeswitch m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-15.. Note the media port has been set to 0 and the rtpmap for G729 is also not correct. On receipt of this bad 200Ok the originator sends a BYE. We are using FS as a B2BUA with bypass_media set to true. Thus IMHO Freeswitch should not be touching the SDP portion of the message and just passing it through. This can reproduce this at will, so I can collect whatever data is nessicary. I have added the sofia debug from the console. Thanks, -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net -------------- next part -------------- tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9bcba58 from (udp/192.168.0.2:5070) has 1303 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9bcba58 (1303 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received INVITE sip:1234567891 at 192.168.0.2:5060 SIP/2.0 (CSeq 811801) nta: INVITE (811801) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9bfa748) called soa_set_params(static::0x9bcca70, ...) called nta_leg_tcreate(0x9c19f60) soa_init_offer_answer(static::0x9bcca70) called soa_set_remote_sdp(static::0x9bcca70, (nil), 0x9ce1541, 286) called nua(0x9bfa748): adding session usage tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 705 bytes of 705 to udp/192.168.0.2:5060 tport_vsend returned 705 nta: sent 100 Trying for INVITE (811801) nua(0x9bfa748): event i_invite 100 Trying nua(0x9bfa748): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x9bcca70, [0xb77b7bac], [0xb77b7ba8], [(nil)]) called nua(0x9bfa748): event i_state 100 Trying nua: nua_application_event: entering 2009-06-04 21:58:34 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/987654321 at sip.evolutiontel.net [6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b] nua: nua_handle_bind: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [received][100] 2009-06-04 21:58:34 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 570753038 570753038 IN IP4 202.xx.xxx.xx s=ENSResip c=IN IP4 202.xx.xxx.xx t=0 0 m=audio 13298 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - 2009-06-04 21:58:34 [DEBUG] sofia.c:3182 sofia_handle_sip_i_state() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_NEW -> CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State INIT 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/987654321 at sip.evolutiontel.net SOFIA INIT 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_INIT -> CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State INIT going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State ROUTING 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/987654321 at sip.evolutiontel.net SOFIA ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/987654321 at sip.evolutiontel.net Standard ROUTING 2009-06-04 21:58:34 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 987654321->1234567891 in context evolutiontel Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unloop] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->global] continue=true Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${network_addr}(192.168.0.2) =~ /^$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net ANTI-Action set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [global] ${numbering_plan}() =~ /^$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set_user(default@${domain_name}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_h_X-ZRTP-On}() =~ /^Y$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_secure_media}() =~ /^true$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_user_agent}(ENSR2.5.4) =~ /^PolycomSound(Point|Station)IP-S(S|P)IP_\d{3,4}-UA\/((3).(\d).(\d).(\d{4}))$/ break=never Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Absolute Condition [global] Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->vmain] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [vmain] destination_number(1234567891) =~ /^121/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->vmain1] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [vmain1] destination_number(1234567891) =~ /^123/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->vmain2] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [vmain2] destination_number(1234567891) =~ /^122/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->ivr_demo] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [ivr_demo] destination_number(1234567891) =~ /^5000$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(1234567891) =~ /^5900$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(1234567891) =~ /^5901$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(1234567891) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(1234567891) =~ /^parking$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(1234567891) =~ /callpark/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(1234567891) =~ /pickup/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->wait] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [wait] destination_number(1234567891) =~ /^wait$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->National_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [National_calls] destination_number(1234567891) =~ /^0(2|3|4|5|7|8|9)[0-9]{8}$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->Special_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [Special_calls] destination_number(1234567891) =~ /^1[3|8][0-9]+$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->International_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (FAIL) [International_calls] destination_number(1234567891) =~ /^0011[0-9]+$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net parsing [evolutiontel->On-Net_calls] continue=false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Regex (PASS) [On-Net_calls] destination_number(1234567891) =~ /^063[0-9]{7}$/ break=on-false Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(execute_on_answer=sched_hangup +${sip_h_x-max-timer} ALLOTED_TIMEOUT) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(sip_cid_type=pid) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(continue_on_fail=79) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(bypass_media=false) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action set(ringback=%(400,200,401,450);%(400,2200,400,450)) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action pre_answer() Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action export(sip_secure_media=true) Dialplan: sofia/internal/987654321 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_ROUTING -> CS_EXECUTE 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State ROUTING going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_EXECUTE 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State EXECUTE 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/987654321 at sip.evolutiontel.net SOFIA EXECUTE 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/987654321 at sip.evolutiontel.net Standard EXECUTE EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(use_profile=nat) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [use_profile]=[nat] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set_user(default at 192.168.0.2) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(bypass_media=true) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(bypass_media=true) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net hash(insert/192.168.0.2-spymap/987654321/6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/987654321/1234567891) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/global/6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(execute_on_answer=sched_hangup +7200 ALLOTED_TIMEOUT) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [execute_on_answer]=[sched_hangup +7200 ALLOTED_TIMEOUT] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(sip_cid_type=pid) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [sip_cid_type]=[pid] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net set(continue_on_fail=79) 2009-06-04 21:58:34 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/987654321 at sip.evolutiontel.net SET [continue_on_fail]=[79] EXECUTE sofia/internal/987654321 at sip.evolutiontel.net bridge({sip_from_uri=sip:987654321 at sip.evolutiontel.net}sofia/internal/1234567891 at 192.168.0.2^0312341234 at 202.xxx.xx.xx:5060) 2009-06-04 21:58:34 [DEBUG] switch_ivr_originate.c:1017 switch_ivr_originate() variable string 0 = [sip_from_uri=sip:987654321 at sip.evolutiontel.net] 2009-06-04 21:58:34 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/1234567891 at 192.168.0.2 [64f91098-4f97-4c17-b95b-0d1693c64f8a] 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:2719 sofia_outgoing_channel() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_INIT 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State INIT 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1234567891 at 192.168.0.2 SOFIA INIT nua: nh_create_handle: entering nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x9d17500): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9d17500) called soa_set_params(static::0x9bf1e20, ...) called soa_set_params(static::0x9bf1e20, ...) called soa_set_user_sdp(static::0x9bf1e20, (nil), 0x9c1d48f, -1) called soa_set_capability_sdp(static::0x9bf1e20, (nil), 0x9c1d48f, -1) called nua(0x9d17500): adding session usage nta_leg_tcreate(0x9bce5c8) soa_init_offer_answer(static::0x9bf1e20) called soa_generate_offer(static::0x9bf1e20, 0) called soa_static_offer_answer_action(0x9bf1e20, soa_generate_offer): called soa_static(0x9bf1e20, soa_generate_offer): generating local description soa_static(0x9bf1e20, soa_generate_offer): upgrade with local description soa_sdp_mode_set(0xb77b7ef4, (nil), ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.125.42.98 (a local address) soa_static(0x9bf1e20, soa_generate_offer): storing local description soa_get_local_sdp(static::0x9bf1e20, [(nil)], [0xb77b7f7c], [0xb77b7f78]) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 1233 bytes of 1233 to udp/192.168.0.2:5060 tport_vsend returned 1233 nta: sent INVITE (115940021) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9cd15d8 for udp/192.168.0.2:5070 (already 0) nua(0x9d17500): call state changed: init -> calling, sent offer soa_get_local_sdp(static::0x9bf1e20, [0xb77b7fa4], [0xb77b7fa0], [(nil)]) called nua(0x9d17500): event i_state INVITE sent nua: nua_application_event: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9d5c900 from (udp/192.168.0.2:5070) has 303 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9d5c900 (303 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 100 Trying for INVITE (115940021) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 1.918 ms tport_release(0x9bb0e28): 0x9cd15d8 by 0x9d5c840 with 0x9d5c900 (preliminary) nua(0x9d17500): sent signal r_invite 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/1234567891 at 192.168.0.2 entering state [calling][0] nua: nua_handle_magic: entering 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State INIT going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_ROUTING 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State ROUTING 2009-06-04 21:58:34 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1234567891 at 192.168.0.2 SOFIA ROUTING 2009-06-04 21:58:34 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State ROUTING going to sleep 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_CONSUME_MEDIA 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State CONSUME_MEDIA 2009-06-04 21:58:34 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State CONSUME_MEDIA going to sleep tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c84f30 from (udp/192.168.0.2:5070) has 711 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9c84f30 (711 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 180 Ringing for INVITE (115940021) nta: 180 Ringing is going to a transaction tport_release(0x9bb0e28): 0x9cd15d8 by 0x9d5c840 with 0x9c84f30 (preliminary) nua(0x9d17500): event r_invite 180 Ringing nua(0x9d17500): call state changed: calling -> proceeding nua(0x9d17500): event i_state 180 Ringing nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/1234567891 at 192.168.0.2 entering state [proceeding][180] 2009-06-04 21:58:34 [NOTICE] sofia.c:3103 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1234567891 at 192.168.0.2! 2009-06-04 21:58:34 [DEBUG] sofia.c:3111 sofia_handle_sip_i_state() sofia/internal/987654321 at sip.evolutiontel.net receive message [RINGING] nua: nua_respond: entering nua(0x9bfa748): sent signal r_respond 2009-06-04 21:58:34 [NOTICE] mod_sofia.c:1422 sofia_receive_message() Ring-Ready sofia/internal/987654321 at sip.evolutiontel.net! 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering nua(0x9bfa748): recv signal r_respond 180 Ringing nua: nua_stack_set_params: entering soa_set_params(static::0x9bcca70, ...) called nua: nua_invite_server_respond: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1132 bytes of 1132 to udp/192.168.0.2:5060 tport_vsend returned 1132 nta: sent 180 Ringing for INVITE (811801) nua(0x9bfa748): call state changed: received -> early nua(0x9bfa748): event i_state 180 Ringing nua: nua_application_event: entering 2009-06-04 21:58:34 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [early][180] nua: nua_handle_magic: entering 2009-06-04 21:58:34 [DEBUG] switch_ivr_originate.c:1768 switch_ivr_originate() sofia/internal/987654321 at sip.evolutiontel.net receive message [RINGING] 2009-06-04 21:58:34 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:34 [NOTICE] switch_ivr_originate.c:1768 switch_ivr_originate() Ring Ready sofia/internal/987654321 at sip.evolutiontel.net! nta: timer set next to 59934 ms tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c855e8 from (udp/192.168.0.2:5070) has 1215 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9c855e8 (1215 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for INVITE (115940021) nta: 200 OK is going to a transaction tport_release(0x9bb0e28): 0x9cd15d8 by 0x9d5c840 with 0x9c855e8 nta: timer shortened to 32000 ms soa_set_remote_sdp(static::0x9bf1e20, (nil), 0x9d149c5, 306) called soa_process_answer(static::0x9bf1e20) called soa_static_offer_answer_action(0x9bf1e20, soa_process_answer): called soa_sdp_mode_set(0x9d197c0, 0x9cec6b0, ""): called soa_static(0x9bf1e20, soa_process_answer): upgrade codecs with remote description soa_static(0x9bf1e20, soa_process_answer): storing local description soa_activate(static::0x9bf1e20, (nil)) called nua(0x9d17500): INVITE: processed SDP answer in 200 OK nua(0x9d17500): event r_invite 200 OK nua: nua_application_event: entering nua: nua_handle_magic: entering soa_activate(static::0x9bf1e20, (nil)) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 621 bytes of 621 to udp/192.168.0.2:5060 tport_vsend returned 621 nta: sent ACK (115940021) to */192.168.0.2:5060 nua(0x9d17500): call state changed: proceeding -> ready, received answer soa_get_remote_sdp(static::0x9bf1e20, [0xb77b7bfc], [0xb77b7bf8], [(nil)]) called soa_get_params(static::0x9bf1e20, ...) called nua(0x9d17500): event i_state 200 OK nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/1234567891 at 192.168.0.2 entering state [ready][200] 2009-06-04 21:58:36 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 4495350 4495350 IN IP4 192.168.0.10 s=- c=IN IP4 60.241.91.137 t=0 0 m=audio 16580 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=direction:active a=oldmediaip:192.168.0.10 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1875 switch_channel_perform_mark_answered() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [NOTICE] sofia.c:3509 sofia_handle_sip_i_state() Channel [sofia/internal/1234567891 at 192.168.0.2] has been answered 2009-06-04 21:58:36 [DEBUG] sofia.c:3522 sofia_handle_sip_i_state() sofia/internal/987654321 at sip.evolutiontel.net receive message [ANSWER] nua: nua_respond: entering nua(0x9bfa748): sent signal r_respond 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [NOTICE] sofia.c:3522 sofia_handle_sip_i_state() Channel [sofia/internal/987654321 at sip.evolutiontel.net] has been answered 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1911 switch_channel_perform_mark_answered() sofia/internal/987654321 at sip.evolutiontel.net execute on answer: sched_hangup(+7200 ALLOTED_TIMEOUT) EXECUTE sofia/internal/987654321 at sip.evolutiontel.net sched_hangup(+7200 ALLOTED_TIMEOUT) 2009-06-04 21:58:36 [DEBUG] switch_scheduler.c:214 switch_scheduler_add_task() Added task 6 switch_ivr_schedule_hangup (6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) to run at 1244123916 nua: nua_handle_magic: entering 2009-06-04 21:58:36 [DEBUG] switch_ivr_originate.c:2024 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/1234567891 at 192.168.0.2] 2009-06-04 21:58:36 [DEBUG] switch_ivr_bridge.c:791 switch_ivr_signal_bridge() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_ivr_bridge.c:792 switch_ivr_signal_bridge() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State EXECUTE going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HIBERNATE 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/987654321 at sip.evolutiontel.net SOFIA HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/987654321 at sip.evolutiontel.net Standard HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HIBERNATE going to sleep nua(0x9d17500): event i_active 200 Call active nua(0x9bfa748): recv signal r_respond 200 OK nua: nua_stack_set_params: entering soa_set_params(static::0x9bcca70, ...) called soa_set_user_sdp(static::0x9bcca70, (nil), 0x9d739a8, -1) called soa_set_capability_sdp(static::0x9bcca70, (nil), 0x9d739a8, -1) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x9bcca70) called soa_static_offer_answer_action(0x9bcca70, soa_generate_answer): called soa_static(0x9bcca70, soa_generate_answer): generating local description soa_static(0x9bcca70, soa_generate_answer): upgrade with remote description soa_static(0x9bcca70, soa_generate_answer): marking rejected media soa_sdp_mode_set(0xb77b7fb4, 0x9c2c578, ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.125.42.98 (a local address) soa_static(0x9bcca70, soa_generate_answer): storing local description soa_activate(static::0x9bcca70, (nil)) called soa_get_local_sdp(static::0x9bcca70, [(nil)], [0xb77b803c], [0xb77b8038]) called tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1421 bytes of 1421 to udp/192.168.0.2:5060 tport_vsend returned 1421 nta: sent 200 OK for INVITE (811801) nta: timer shortened to 500 ms nua(0x9bfa748): call state changed: early -> completed, sent answer soa_get_local_sdp(static::0x9bcca70, [0xb77b8124], [0xb77b8120], [(nil)]) called soa_get_params(static::0x9bcca70, ...) called nua(0x9bfa748): event i_state 200 OK 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HIBERNATE 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/1234567891 at 192.168.0.2 SOFIA HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/1234567891 at 192.168.0.2 Standard HIBERNATE 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HIBERNATE going to sleep nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [completed][200] nua: nua_handle_magic: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9ce6898 from (udp/192.168.0.2:5070) has 479 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9ce6898 (479 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received ACK sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 811801) nta: ACK (811801) is going to INVITE (811801) nua: process_ack_or_cancel: entering soa_clear_remote_sdp(static::0x9bcca70) called nua(0x9bfa748): event i_ack 200 OK nua(0x9bfa748): call state changed: completed -> ready nua(0x9bfa748): event i_state 200 OK nua(0x9bfa748): event i_active 200 Call active nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [ready][200] nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9ce6898 from (udp/192.168.0.2:5070) has 479 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9ce6898 (479 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 811802) nta: canonizing sip:mod_sofia at 192.168.0.2:5070 with contact nta: BYE (811802) going to existing leg nua: nua_stack_process_request: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 593 bytes of 593 to udp/192.168.0.2:5060 tport_vsend returned 593 nta: sent 200 OK for BYE (811802) nua(0x9bfa748): event i_bye 200 Session Terminated nua(0x9bfa748): removing session usage nua(0x9bfa748): call state changed: ready -> terminated nua(0x9bfa748): event i_state 200 Session Terminated nua(0x9bfa748): event i_terminated 200 Session Terminated soa_destroy(static::0x9bcca70) called nta_leg_destroy(0x9c19f60) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-04 21:58:36 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/987654321 at sip.evolutiontel.net entering state [terminated][200] 2009-06-04 21:58:36 [NOTICE] sofia.c:3599 sofia_handle_sip_i_state() Hangup sofia/internal/987654321 at sip.evolutiontel.net [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/987654321 at sip.evolutiontel.net [KILL] 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua(0x9bfa748): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua: nua_application_event: entering nua(0x9bfa748): event i_terminated dropped 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_HANGUP 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HANGUP 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/987654321 at sip.evolutiontel.net Overriding SIP cause 480 with 200 from the other leg 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/987654321 at sip.evolutiontel.net hanging up, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [NOTICE] switch_ivr_bridge.c:712 signal_bridge_on_hangup() Hangup sofia/internal/1234567891 at 192.168.0.2 [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-04 21:58:36 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/1234567891 at 192.168.0.2 [KILL] 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/987654321 at sip.evolutiontel.net Standard HANGUP, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State HANGUP going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_HANGUP -> CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/987654321 at sip.evolutiontel.net [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) Running State Change CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/987654321 at sip.evolutiontel.net) State REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/987654321 at sip.evolutiontel.net Standard REPORTING, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/987654321 at sip.evolutiontel.net) State REPORTING going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/987654321 at sip.evolutiontel.net) State Change CS_REPORTING -> CS_DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 7 (sofia/internal/987654321 at sip.evolutiontel.net) Locked, Waiting on external entities 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/987654321 at sip.evolutiontel.net) Ended 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/987654321 at sip.evolutiontel.net [CS_DESTROY] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/987654321 at sip.evolutiontel.net) State DESTROY 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/987654321 at sip.evolutiontel.net SOFIA DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/987654321 at sip.evolutiontel.net Standard DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/987654321 at sip.evolutiontel.net) State DESTROY going to sleep nua(0x9bfa748): recv signal r_destroy nta_leg_destroy((nil)) 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_HANGUP 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HANGUP 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/1234567891 at 192.168.0.2 Overriding SIP cause 480 with 200 from the other leg 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/1234567891 at 192.168.0.2 hanging up, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:394 sofia_on_hangup() Sending BYE to sofia/internal/1234567891 at 192.168.0.2 nua: nua_bye: entering nua(0x9d17500): sent signal r_bye 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1234567891 at 192.168.0.2 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State HANGUP going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1234567891 at 192.168.0.2 [BREAK] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) Running State Change CS_REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1234567891 at 192.168.0.2) State REPORTING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/1234567891 at 192.168.0.2 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1234567891 at 192.168.0.2) State REPORTING going to sleep 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1234567891 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 8 (sofia/internal/1234567891 at 192.168.0.2) Locked, Waiting on external entities 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 8 (sofia/internal/1234567891 at 192.168.0.2) Ended 2009-06-04 21:58:36 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1234567891 at 192.168.0.2 [CS_DESTROY] 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1234567891 at 192.168.0.2) State DESTROY 2009-06-04 21:58:36 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/1234567891 at 192.168.0.2 SOFIA DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/1234567891 at 192.168.0.2 Standard DESTROY 2009-06-04 21:58:36 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1234567891 at 192.168.0.2) State DESTROY going to sleep nua(0x9d17500): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x9bf1e20, ...) called soa_terminate(static::0x9bf1e20) called soa_init_offer_answer(static::0x9bf1e20) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 871 bytes of 871 to udp/192.168.0.2:5060 tport_vsend returned 871 nta: sent BYE (115940022) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9d19bc8 for udp/192.168.0.2:5070 (already 0) 2009-06-04 21:58:36 [DEBUG] switch_scheduler.c:138 task_thread_loop() Deleting task 6 switch_ivr_schedule_hangup (6dd5c26f-8cfc-4876-a538-ed5f5e03cc1b) tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c84f30 from (udp/192.168.0.2:5070) has 449 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9c84f30 (449 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for BYE (115940022) nta: 200 OK is going to a transaction nta_outgoing: RTT is 97.734 ms tport_release(0x9bb0e28): 0x9d19bc8 by 0x9bb6ef8 with 0x9c84f30 nua(0x9d17500): event r_bye 200 OK nua(0x9d17500): call state changed: terminating -> terminated nua(0x9d17500): event i_state 200 to BYE nua(0x9d17500): event i_terminated 200 to BYE nua(0x9d17500): removing session usage soa_destroy(static::0x9bf1e20) called nta_leg_destroy(0x9bce5c8) nua: terminated session 0x9d17500 nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x9d17500): sent signal r_destroy nua: nua_application_event: entering nua(0x9d17500): event i_terminated dropped nua(0x9d17500): recv signal r_destroy nta_leg_destroy((nil)) nta: timer set next to 4634 ms tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9ce6aa0 from (udp/192.168.0.2:5070) has 479 bytes, veclen = 1 tport_deliver(0x9bb0e28): msg 0x9ce6aa0 (479 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 811802) nta: BYE (811802) going to existing BYE transaction nta: re-received BYE request, retransmitting 200 reply tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 593 bytes of 593 to udp/192.168.0.2:5060 tport_vsend returned 593 nta: timer I fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0xb77b81cc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 107 ms nta: timer K fired, terminate BYE (115940022) outgoing_reclaim_all((nil), (nil), 0xb77b81c8) nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/2 term, 1/3 free nta: timer set next to 26750 ms /exit [root at sip01 bin]# From anthony.minessale at gmail.com Thu Jun 4 05:49:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 07:49:41 -0500 Subject: [Freeswitch-users] Error causing freeswitch to crash In-Reply-To: References: <1CEC9ECBBE2B49FB88E474EB2F8F9BD5@D810> Message-ID: <191c3a030906040549y2ec5fe8aica40122032057715@mail.gmail.com> also make sure you have a clean update to SVN trunk before re-testing. On Thu, Jun 4, 2009 at 3:59 AM, Michael Jerris wrote: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > Please attempt to reproduce this issue with trunk with crash protection > disabled, and if you are able please file a jira with a backtrace of the > crash > > Mike > > On Jun 4, 2009, at 4:23 AM, Andy Ayers wrote: > > Hi, > > Every few days I'm getting this error which is causing Freeswitch to crash. > Can anyone tell me what may be causing this or how to prevent it? > > 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 > handle_fatality() Caught signal 11 for unmapped thread! > > Many thanks > Andy > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/be334034/attachment-0002.html From rupa at rupa.com Thu Jun 4 05:54:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 4 Jun 2009 07:54:24 -0500 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: <200906041402.43704.yivzhenko@mksat.net> References: <200906041402.43704.yivzhenko@mksat.net> Message-ID: The API mthod of setting the callerid is really for testing purposes. Since you have no callerid from the FS command line interface I added a parameter you can pass to specify the CID. When in the dialplan you actually have the CID, so none is used. mod_lcr is just grabbing the callerid on the channel. I assme this is not the base callerid you want passed to the sip provider? On Thu, Jun 4, 2009 at 6:02 AM, Yuriy Ivzhenko wrote: > Hello. > > I have another problem with setting caller id with cid field. > > command lcr has parameter [caller_id]. > > and if i call lcr from the commandline it works fine, > > but if i call lcr from dialplan it ignores this parameter and use original > caller id from caller. > > my cid field = /^(.*)$/999$1/ > > If i call lcr from the commandline > > lcr 444555 default 11111 > > lcr return origination_caller_id_number=99911111 > > but if i call lcr from dialplan > > > > lcr return origination_caller_id_number=9991002 > > 1002 is a original caller id number > > There is a way to use this parameter from dialplan? > > Yuriy > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/2e1d8f0c/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 4 05:55:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 07:55:52 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: <191c3a030906040555k1168c03an4330235ab8b69320@mail.gmail.com> Did you try what he said? The new nat busting stuff will allow you to use just the internal profile for everything. you may need to join irc and ask bkw for the proper config options. On Thu, Jun 4, 2009 at 4:41 AM, Jason White wrote: > Michael Jerris wrote: > > Can you please re-test with current svn trunk. we added some new nat > > busting code yesterday that may assist with this. You will need to > > specify the new > > param in the sofia profile (see > http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 > > ) > > If I apply this to the external profile (using it on the internal profile > has > no effect), the Via headers now receive the external IP address, i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d880d079/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 05:59:40 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 22:59:40 +1000 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: This is the one you need to change c=IN IP4 192.168.0.2. o= is just the owner whereas c= is the connection IP. Try rtp-ip in your sip profile Having said that Warning: 301 203.2.134.1 "invalid transport IP address" isnt 203.2.134.1 the address it is complaining about? On Thu, Jun 4, 2009 at 7:41 PM, Jason White wrote: > Michael Jerris wrote: >> Can you please re-test with current svn trunk. ?we added some new nat >> busting code yesterday that may assist with this. ?You will need to >> specify the new > ?> param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 >> ? ) > > If I apply this to the external profile (using it on the internal profile has > no effect), the Via headers now receive the external IP address, i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From anthony.minessale at gmail.com Thu Jun 4 05:59:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 07:59:13 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: Message-ID: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> you should have also turned in the sip trace sofia profile internal siptrace on On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke wrote: > Hi All, > > Looking for some debugging tips and comments on what might be causing > the media port in the 200OK ( Answer message) to be set to 0 by > freeswitch. Essentially it looks like data might be getting trampled > somehow. > > Portion of 200OK going into Freeswitch > m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 > NSE/8000..a= > fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 > 0-15..a=ptime:20..a=sendrecv.. > > Portion of 200OK coming out of Freeswitch > m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 > NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event > /8000..a=fmtp:101 0-15.. > > Note the media port has been set to 0 and the rtpmap for G729 is also > not correct. On receipt of this bad 200Ok the originator sends a BYE. > > We are using FS as a B2BUA with bypass_media set to true. Thus IMHO > Freeswitch should not be touching the SDP portion of the message and > just passing it through. > > This can reproduce this at will, so I can collect whatever data is > nessicary. I have added the sofia debug from the console. > > Thanks, > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d8a4ddaa/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 06:09:18 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 23:09:18 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> Message-ID: I have traces using ngrep, however if you want to see it all in one file I will collect tommorow. Regards, On Thu, Jun 4, 2009 at 10:59 PM, Anthony Minessale wrote: > you should have also turned in the sip trace > sofia profile internal siptrace on > > > On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke wrote: >> >> Hi All, >> >> Looking for some debugging tips and comments on what might be causing >> the media port in the 200OK ( Answer message) to be set to 0 by >> freeswitch. ?Essentially it looks like data might be getting trampled >> somehow. >> >> Portion of 200OK going into Freeswitch >> m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 >> NSE/8000..a= >> ?fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >> 0-15..a=ptime:20..a=sendrecv.. >> >> Portion of 200OK coming out of Freeswitch >> m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 >> NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event >> ?/8000..a=fmtp:101 0-15.. >> >> Note the media port has been set to 0 and the rtpmap for G729 is also >> not correct. ?On receipt of this bad 200Ok the originator sends a BYE. >> >> We are using FS as a B2BUA with bypass_media set to true. ?Thus IMHO >> Freeswitch should not be touching the SDP portion of the message and >> just passing it through. >> >> This can reproduce this at will, so I can collect whatever data is >> nessicary. ?I have added the sofia debug from the console. >> >> Thanks, >> -- >> Jim Burke >> Director Evolutiontel. >> http://www.evolutiontel.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From jim at evolutiontel.net Thu Jun 4 06:10:23 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 4 Jun 2009 23:10:23 +1000 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: Apologies that should have been ext-rtp-ip Regards, On Thu, Jun 4, 2009 at 10:59 PM, Jim Burke wrote: > This is the one you need to change c=IN IP4 192.168.0.2. ?o= is just > the owner whereas c= is the connection IP. ?Try rtp-ip in your sip > profile > > Having said that Warning: 301 203.2.134.1 "invalid transport IP > address" isnt 203.2.134.1 the address it is complaining about? > > On Thu, Jun 4, 2009 at 7:41 PM, Jason White wrote: >> Michael Jerris wrote: >>> Can you please re-test with current svn trunk. ?we added some new nat >>> busting code yesterday that may assist with this. ?You will need to >>> specify the new >> ?> param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 >>> ? ) >> >> If I apply this to the external profile (using it on the internal profile has >> no effect), the Via headers now receive the external IP address, i.e., the >> publicly routable address provided by the ISP. >> >> However, the session description still has the private address in it, which >> causes the remote end to issue the following: >> SIP/2.0 488 Invalid Session Description >> Warning: 301 203.2.134.1 "invalid transport IP address" >> >> I assume (but may be wrong - full traces can be provided if necessary) that >> the problem is in the SDP that we're sending out: >> o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 192.168.0.2 >> >> Can this be fixed up? >> >> Thanks. >> >> Jason. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From yivzhenko at mksat.net Thu Jun 4 06:15:15 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 4 Jun 2009 16:15:15 +0300 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: References: <200906041402.43704.yivzhenko@mksat.net> Message-ID: <200906041615.16453.yivzhenko@mksat.net> On Thursday 04 June 2009 15:54:24 Rupa Schomaker wrote: > The API mthod of setting the callerid is really for testing purposes. > Since you have no callerid from the FS command line interface I added a > parameter you can pass to specify the CID. > > When in the dialplan you actually have the CID, so none is used. > > mod_lcr is just grabbing the callerid on the channel. I assme this is not > the base callerid you want passed to the sip provider? > Yes, there is internal callerid on the channel. I have external (PSTN) number, associated with this internal number. And i need to pass them to provider. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/54289581/attachment-0002.html From durk.debeer at isp.solcon.nl Thu Jun 4 02:53:05 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Thu, 04 Jun 2009 11:53:05 +0200 Subject: [Freeswitch-users] Attendant transfer problem with a Cisco phone and Freeswitch. Message-ID: I've come across a problem when using Cisco phones as sip-clients on a Freeswitch system. The problem that arises has to do whit which RFC the Cisco phones are following. Best thing to do at this moment is to point to the Tech-invite website http://www.tech-invite.com/. The examples on that site are more explanatory to the problem than I am ever are able to provide. Ok if you take a look at the link SIP Service Examples there are 19 examples of how RFC 5359 describes how call signalling should flow. If you take a careful look at example 5 (attendant transfer) you will discover that before there is the transfer the station the transfer will go to is put on hold. Bob is transfering to Carol so she is being put on hold by Bob, signals 12 to 14. Now if you take a look at the RFC involved in transfering calls, to be found on the main site link SIP-Topics (to the left) and then following the link Call Transfer (middle window second line in yellow section) you'll find three RFC regarding to transfering calls. In none of these am I able to find this putting on hold as is scribed in RFC 5359 so it seems reasonably to assume that this 'putting on hold' in not mandatory. Here now arises the source of my problem, the Cisco phones are using RFC 5359 when attempting a attendant transfer. When now this signalling is to flow through Freeswitch it puts the station where the transfer is going to on hold as in the prior example is happening to Carol. Freeswitch the connects the MOH stream to this station. Seems a logical thing to do if your assume the putting on hold is not mandatory for the transfer. When now the original station (prior example Bob) is sending the refer thins go bad. Freeswitch is not sending new invites or other signaling to the stations. It is only processing the byes from the station that preforms the transfer (prior example Bob). If I now break the MOH stream on the freeswitch cli all goes well meaning that all invites and other signalling is flowing correctly through Freeswitch. Did anyone out there have the same problem or better yet have a fix for it?. ? Kind regards, Durk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0e30f0c0/attachment.html From rupa at rupa.com Thu Jun 4 06:57:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 4 Jun 2009 08:57:52 -0500 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: <200906041615.16453.yivzhenko@mksat.net> References: <200906041402.43704.yivzhenko@mksat.net> <200906041615.16453.yivzhenko@mksat.net> Message-ID: Update to at least rev 13611 and mod_lcr will check "effective_caller_id_number" prior to the real callerid on the channel. This works in the same way that bridge does so no surprises. On Thu, Jun 4, 2009 at 8:15 AM, Yuriy Ivzhenko wrote: > On Thursday 04 June 2009 15:54:24 Rupa Schomaker wrote: > > > The API mthod of setting the callerid is really for testing purposes. > > > Since you have no callerid from the FS command line interface I added a > > > parameter you can pass to specify the CID. > > > > > > When in the dialplan you actually have the CID, so none is used. > > > > > > mod_lcr is just grabbing the callerid on the channel. I assme this is not > > > the base callerid you want passed to the sip provider? > > > > > Yes, there is internal callerid on the channel. I have external (PSTN) > number, associated with this internal number. And i need to pass them to > provider. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/6e38ee9a/attachment-0002.html From brian at freeswitch.org Thu Jun 4 07:07:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:07:26 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: <079FC3ED-1F71-4257-A651-FCA66B7C8EB4@freeswitch.org> Yah make sure your ext-rtp-ip is set or hop on IRC and let me help you. /b On Jun 4, 2009, at 4:41 AM, Jason White wrote: > If I apply this to the external profile (using it on the internal > profile has > no effect), the Via headers now receive the external IP address, > i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in > it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if > necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 > 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/64944d25/attachment-0002.html From bruce.mcalister at blueface.ie Thu Jun 4 07:10:11 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Thu, 04 Jun 2009 15:10:11 +0100 Subject: [Freeswitch-users] Solaris 10 build fails with "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications require the use of c99" In-Reply-To: <6297FD9D-142A-4E58-B4AC-6294652EAAF6@jerris.com> References: <4A262331.2080104@blueface.ie> <4A2646BF.7090607@blueface.ie> <20090603101746.GA3854@jdc.jasonjgw.net> <4A266025.1070505@blueface.ie> <4A26D26C.8080107@blueface.ie> <6297FD9D-142A-4E58-B4AC-6294652EAAF6@jerris.com> Message-ID: <4A27D5C3.4060209@blueface.ie> Hi Mike, As per request, here is the build status of 1.0.4preX for me: 1.0.4pre2 compile successful 1.0.4pre3 compile successful 1.0.4pre4 compile successful 1.0.4pre5 compile successful 1.0.4pre6 compile successful 1.0.4pre7 compile successful 1.0.4pre8 compile fails I have used the same spec file for each build so the build environment is identical for each. I have filed a jira at: http://jira.freeswitch.org/browse/FSBUILD-169 Thanks Bruce Michael Jerris wrote: > if you could nail down a specific svn revision that causes this issue > and file a jira at http://jira.freeswitch.org that would be a big help > in resolving this issue. > > Mike > > On Jun 3, 2009, at 3:43 PM, Bruce McAlister wrote: > >> Hi All, >> >> Any pointers or suggestions on this issue would be greatly >> appreciated. >> >> PS: I tried compiling several version of FreeSWITCH to see if I >> encounter the same issue, I have varying results. Version 1.0 compiles >> fine, version 1.0.2 fails and 1.0.3RC1/1.0.3 builds fine. But this >> error >> message is new in version 1.0.4preX. I've not tried any older >> pre-releases of 1.0.4 though. >> >> Thanks >> Bruce >> >> Thanks >> Bruce >>> /bin/bash >>> /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> libtool >>> --silent --mode=compile /usr/sfw/bin/gcc -m32 -DHAVE_CONFIG_H >>> -DSOLARIS2=10 -D_POSIX_PTHREAD_SEMANTICS -D_REENTRANT >>> -D_LARGEFILE64_SOURCE -I./include >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> include/arch/unix >>> -I./include/arch/unix >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> include >>> -o locks/unix/thread_mutex.lo -c locks/unix/thread_mutex.c && touch >>> locks/unix/thread_mutex.lo >>> In file included from >>> /usr/sfw/lib/gcc/i386-pc-solaris2.10/3.4.3/include/sys/types.h:27, >>> from ./include/apr.h:113, >>> from >>> /export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr/ >>> include/arch/unix/apr_arch_thread_mutex.h:24, >>> from locks/unix/thread_mutex.c:17: >>> /usr/include/sys/feature_tests.h:336:2: #error "Compiler or options >>> invalid; UNIX 03 and POSIX.1-2001 applications require the use >>> of c99" >>> make[2]: *** [locks/unix/thread_mutex.lo] Error 1 >>> make[2]: Leaving directory >>> `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' >>> make[1]: *** [all-recursive] Error 1 >>> make[1]: Leaving directory >>> `/export/home/user/packages/BUILD/freeswitch-1.0.4pre8/libs/apr' >>> make: *** [libs/apr/libapr-1.la] Error 2 >>> >>> In all cases I have started the build from the beginning, whereby I >>> remove and re-extract the 1.0.4pre8 tarball. I've tried with just a >>> configure and also a bootstrap/configure, but I end up with the same >>> error (except when I change the compiler to Sun Studio 12's c99). >>> >>> Is GCC 3.4.3 too old to use to build this version of freeswitch? >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Jun 4 07:13:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:13:29 -0500 Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: <8548096.287831244101967346.JavaMail.root@zimbra.tng.de> References: <8548096.287831244101967346.JavaMail.root@zimbra.tng.de> Message-ID: Sorry but this type of trace is impossible to read. I want raw pcap if possible. /b On Jun 4, 2009, at 2:52 AM, Rudolf Denert wrote: > Ok, > > here is the SIP trace. If you need more, just tell me and I will > send them. The RTP trace you already have, haven't you? > > Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3621de5a/attachment-0002.html From brian at freeswitch.org Thu Jun 4 07:16:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:16:20 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: <6DE16413-98FF-4BC3-8067-A7C27081D989@freeswitch.org> The private address in the SDP's o= line is fine. If the far end is rejecting that then something is broken on their side. The c= line is all that matters in this case... For the record we do provide the complete correct SDP to sofia but the lib overrides the o= line and replaces it. I have narrowed down the exact lines of code that causes this in soa.c in sofia... I have emailed the author of the lib to ask why and how to keep sofia from messing with that o= line. /b On Jun 4, 2009, at 4:41 AM, Jason White wrote: > If I apply this to the external profile (using it on the internal > profile has > no effect), the Via headers now receive the external IP address, > i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in > it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if > necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 > 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/03c342ba/attachment-0002.html From yivzhenko at mksat.net Thu Jun 4 07:21:13 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 4 Jun 2009 17:21:13 +0300 Subject: [Freeswitch-users] mod_lcr caller id In-Reply-To: References: <200906041402.43704.yivzhenko@mksat.net> <200906041615.16453.yivzhenko@mksat.net> Message-ID: <200906041721.13791.yivzhenko@mksat.net> Impressive!!! :) On Thursday 04 June 2009 16:57:52 Rupa Schomaker wrote: > Update to at least rev 13611 and mod_lcr will check > "effective_caller_id_number" prior to the real callerid on the channel. > This works in the same way that bridge does so no surprises. > > On Thu, Jun 4, 2009 at 8:15 AM, Yuriy Ivzhenko wrote: > > On Thursday 04 June 2009 15:54:24 Rupa Schomaker wrote: > > > The API mthod of setting the callerid is really for testing purposes. > > > > > > Since you have no callerid from the FS command line interface I added a > > > > > > parameter you can pass to specify the CID. > > > > > > > > > > > > When in the dialplan you actually have the CID, so none is used. > > > > > > > > > > > > mod_lcr is just grabbing the callerid on the channel. I assme this is > > > not > > > > > > the base callerid you want passed to the sip provider? > > > > Yes, there is internal callerid on the channel. I have external (PSTN) > > number, associated with this internal number. And i need to pass them to > > provider. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/f6b62921/attachment-0002.html From god.nirvana at gmail.com Thu Jun 4 07:19:54 2009 From: god.nirvana at gmail.com (god.nirvana) Date: Thu, 4 Jun 2009 22:19:54 +0800 Subject: [Freeswitch-users] busy tone detect issue Message-ID: <200906042219500623349@gmail.com> hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 openzap.conf: [span zt FXO1] name => OpenZAP-FXO1 number => 1 fxo-channel => 1 [span zt FXO2] name => OpenZAP-FXO2 number => 2 fxo-channel => 2 [span zt FXO3] name => OpenZAP-FXO3 number => 3 fxo-channel => 3 [span zt FXO4] name => OpenZAP-FXO4 number => 4 fxo-channel => 4 openzap.conf.xml : when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/47dd5d21/attachment-0002.html From regs at kinetix.gr Thu Jun 4 07:32:22 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 04 Jun 2009 17:32:22 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior Message-ID: <4A27DAF6.30005@kinetix.gr> In the process of trying to use Freeswitch in a production environment I conducted a number of performance tests using various servers. It was then that I noticed some strange behavior from FS. When I stripped down the scenario I was using to a simple bridge scenario, I stumbled upon a strange behavior. The scenario as I stated is quite simple. |---------| |---------| | |------- Call from sipp------> | | | sipp | | FS | | | <------ Call back to sipp----| | |---------| |---------| I did not use an RTP stream for my calls just to test the signaling alone. The sipp scenario is the standard uac.xml scenario that can be found integrated to sipp with the following options : Test FS 1: sipp :5060 -s 55555555 -i -mi -ci -r 10 -d 5000 -l 100 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~(35-60) % (35 during the generation of new calls, 60 during the -l limit imposed by the test) Note : 985 of them had a duration (billsec) of 10 and 15 of them had a duration of 11. I tried raising the call rate and limit... Test FS 1: sipp :5060 -s 55555555 -i -mi -ci -r 20 -d 5000 -l 200 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~0-30 % (0 during the generation of new calls, 30 during the -l limit imposed by the test) THIS IS WHAT MAKES ME WONDER : The distribution of the durations (billsec - not complete durations) : 183 calls with 10 secs billed duration 110 calls with 11 secs billed duration 238 calls with 12 secs billed duration 447 calls with 13 secs billed duration 22 calls with 14 secs billed duration The sipp scenario is simple "hangup the phone after 10 secs". So, why am I seeing these? Of course that has something to do with the stress the machine has been put through during the second test. But I can see it happening to less stressful conditions (i.e. 15 calls per second) to a smaller extend. I captured one of these calls and verified that when the sipp client hangs up exactly 10 secs after the call start, FS receives the BYE and replies with 200 OK. BUT it does not hang the second leg in a timely manner i.e. it sends a BYE to the sipp server side 1-4 seconds AFTER that. That explains the 11, 12, 13, 14 secs durations seen on the second test. What is more interesting is that I would expect to see in the CDRs the first and second leg to have different durations (since the a leg BYE was received and aknowledged by FS in the correct time) i.e. 10 and 14 secs accordingly. But what I get is the same duration for both legs (14 secs). This in my opinion is very dangerous on production environments (you get charged by your provider more seconds that you charge your clients - or - you falsely charge your clients with bigger durations although they hunged up corectly (and you acknowledged it)). NOTE No 1 : All the performance recommendations found in the wiki has been applied. In fact only the essential modules that could make this scenario work were loaded. NOTE No 2 : I tried using asterisk (as a point of reference - don't get me wrong - I am not trying to start a flame war here). And it succeeded doing on the same hardware 60 calls/sec with a channel limit of 400 sim. calls using only 50% of the cpu (maximum). No under any circumstances I have seen the behavior above (this inability to hang call legs in a timely manner). Even when I pushed asterisk to the limits (80 calls per second 600 max call limit) and it started failing on some calls it never failed to hangup the calls for both legs on exactly 10 secs. NOTE No 3 : As you can tell I was using a very small machine for my tests. When I moved the same tests to larger installations (Quad Core Opterons and Xeons) I got proportional results to the above. NOTE No 4 : The tests were performed in a LAN environment and since there was no RTP involved I think there were no bandwidth issues there. NOTE No 5 : The tests were performed using numerous SVN versions (latest : 13610), the stable version and the 1.0.4pre8 version. NOTE No 6 : Using the -hp switch made no noticeable change in behavior. I am not trying to complain for FS's performance (far from it). I am just somewhat disappointed seeing it performing in such a strange manner when under stress. I would prefer a design that drops the calls after a certain threshold than a design that incorrectly handles them all (I am aware of the max sessions per second in switch.conf.cml - but I am starting to see this behavior even with the cpu idling at about 80%). I don't know if anyone else had the same experience when testing Freeswitch. I can happily supply with all the test details (config files, captures etc) to all interested parties. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From brian at freeswitch.org Thu Jun 4 07:40:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 09:40:59 -0500 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A27DAF6.30005@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> Message-ID: <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> On Jun 4, 2009, at 9:32 AM, Apostolos Pantsiopoulos wrote: > NOTE No 1 : All the performance recommendations found in the wiki has > been applied. In fact only the essential modules that could make this > scenario work > were loaded. What are you testing against? What OS, Hardware, Distro and such? > NOTE No 2 : I tried using asterisk (as a point of reference - don't > get > me wrong - I am not trying to start a flame war here). And it > succeeded > doing on the same hardware 60 calls/sec with a channel limit of 400 > sim. calls using only 50% of the cpu (maximum). No under any > circumstances I have seen the behavior above (this inability to hang > call legs in a timely manner). Even when I pushed asterisk to the > limits > (80 calls per second 600 max call limit) and it started failing on > some > calls it never failed to hangup the calls for both legs on exactly > 10 secs. Load testing is a science and you can do it wrong most of the time unless you know exactly what you're doing. If you're going against the default dialplan its heavy and not something I would load test against. > NOTE No 3 : As you can tell I was using a very small machine for my > tests. When I moved the same tests to larger installations (Quad Core > Opterons and Xeons) I got proportional results to the above. What are you testing on now? Hope its 64bit. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3389eb15/attachment-0002.html From regs at kinetix.gr Thu Jun 4 08:01:54 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 04 Jun 2009 18:01:54 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> Message-ID: <4A27E1E2.6070905@kinetix.gr> Brian West wrote: > > On Jun 4, 2009, at 9:32 AM, Apostolos Pantsiopoulos wrote: > >> NOTE No 1 : All the performance recommendations found in the wiki has >> been applied. In fact only the essential modules that could make this >> scenario work >> were loaded. > > What are you testing against? What OS, Hardware, Distro and such? The small server tests were made on a 5-year old PC (32 bit, 3 Ghz P4, Cetnos 5.3). The large server 1 : Quad-Core AMD Opteron(tm) Processor 2350 HE (64 bit, Centos 5.3) The large server 2 : Dual-Core AMD Opteron(tm) Processor 2214 HE (64 bit, Centos 5.3) The large server 3 : Intel(R) Xeon(R) CPU E5345 @ 2.33GHz (32 bit, Centos 5.3) > >> NOTE No 2 : I tried using asterisk (as a point of reference - don't get >> me wrong - I am not trying to start a flame war here). And it succeeded >> doing on the same hardware 60 calls/sec with a channel limit of 400 >> sim. calls using only 50% of the cpu (maximum). No under any >> circumstances I have seen the behavior above (this inability to hang >> call legs in a timely manner). Even when I pushed asterisk to the limits >> (80 calls per second 600 max call limit) and it started failing on some >> calls it never failed to hangup the calls for both legs on exactly 10 >> secs. > > Load testing is a science and you can do it wrong most of the time > unless you know exactly what you're doing. If you're going against the > default dialplan its heavy and not something I would load test against. The dialplan : I think it is the simplest that can be used in this scenario. > > >> NOTE No 3 : As you can tell I was using a very small machine for my >> tests. When I moved the same tests to larger installations (Quad Core >> Opterons and Xeons) I got proportional results to the above. > > What are you testing on now? Hope its 64bit. Most of the platforms were 64 bit (although the results that I posted were from the small 32-bit server, the results from the 64-bit servers were proportional to those). In other words we needed a large call/sec rate for the high end servers but in any case the same phenomenon occured at around 60% idle cpu. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From msc at freeswitch.org Thu Jun 4 09:38:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 09:38:01 -0700 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A27E1E2.6070905@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> Message-ID: <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> > > > The dialplan : > > > > > > > > You forgot the parens around .* It should be expression="^(.*)$" if you plan to use $1 later in the extension... > > > data="absolute_codec_string=PCMA"/> > data="sofia/gateway/sipp01/$1"/> ... like here ^^^^^^^ :) -MC > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/a30ef693/attachment-0002.html From evilla at chipoly.com Thu Jun 4 09:45:32 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Thu, 4 Jun 2009 10:45:32 -0600 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup Message-ID: <01ac01c9e533$e0b79430$a226bc90$@com> Hello guys, I have the A400 card and I'm using: wanpipe-3.5.2 driver zaptel-1.4.11 oslec-0.2 freeswitch 1.0.3 CentOS 5.2 x86_64 kernel 2.6.18-12.128.1.10el5 I had my system working with no problems. After I did shutdown/restart, there was a problem loading wanrouter. Please look at this error log: http://pastebin.freeswitch.org/9246 Then I did reinstal zaptel and oslec again, and then wanrouter got to start with no problem. The only issue was that freeswitch got core dumps when starting, so could not get working again the system. For testing purposes I shutdown / restart the box again and got the same error... Any ideas where to look at? ChiPoLy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/7037385c/attachment-0002.html From brian at freeswitch.org Thu Jun 4 09:48:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 11:48:37 -0500 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup In-Reply-To: <01ac01c9e533$e0b79430$a226bc90$@com> References: <01ac01c9e533$e0b79430$a226bc90$@com> Message-ID: Please report your bugs to jira.freeswitch.org... if you have a segfault then thats where it belongs. Thanks, Brian On Jun 4, 2009, at 11:45 AM, Ing. Edwin Villarreal wrote: > Then I did reinstal zaptel and oslec again, and then wanrouter got > to start with no problem. The only issue was that freeswitch got > core dumps when starting, so could not get working again the system. > > For testing purposes I shutdown / restart the box again and got the > same > error... > > Any ideas where to look at? > > ChiPoLy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/1017b04f/attachment-0002.html From larclap at yahoo.com Thu Jun 4 09:59:00 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 4 Jun 2009 09:59:00 -0700 Subject: [Freeswitch-users] Missing lines copying data from console to vi Message-ID: <079f01c9e535$c1fed2e0$45fc78a0$@com> I want to copy the results of a siptrace captured on the fs console to a file. The console is running on a Gnome terminal. I highlight the text I want to copy in the fs console, open a vi session in insert mode, and paste the text. However, the text is not pasted as I copied it - it is missing characters/lines. I know I am doing something wrong. Is there another way to save siptraces to a file? Redirection doesn't work. sofia profile internal siptrace on is the command I use. Thanks Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/9addf86b/attachment-0002.html From evilla at chipoly.com Thu Jun 4 10:03:47 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Thu, 4 Jun 2009 11:03:47 -0600 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup In-Reply-To: References: <01ac01c9e533$e0b79430$a226bc90$@com> Message-ID: <01e901c9e536$6d8aafd0$48a00f70$@com> Thanks Brian. After commenting mod_openzap in modules.conf.xml I get to run FS with no problem. Problem is with Zaptel / Wanpipe interface. L Ing. Edwin Villarreal World Net Commerce SA CV De: Brian West [mailto:brian at freeswitch.org] Enviado el: jueves, 04 de junio de 2009 10:49 a.m. Para: freeswitch-users at lists.freeswitch.org Asunto: Re: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup Please report your bugs to jira.freeswitch.org... if you have a segfault then thats where it belongs. Thanks, Brian On Jun 4, 2009, at 11:45 AM, Ing. Edwin Villarreal wrote: Then I did reinstal zaptel and oslec again, and then wanrouter got to start with no problem. The only issue was that freeswitch got core dumps when starting, so could not get working again the system. For testing purposes I shutdown / restart the box again and got the same error... Any ideas where to look at? ChiPoLy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/f6fb9b3e/attachment-0002.html From brian at freeswitch.org Thu Jun 4 10:06:07 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 12:06:07 -0500 Subject: [Freeswitch-users] Zaptel / Wanpipe broken after shutdown/startup In-Reply-To: <01e901c9e536$6d8aafd0$48a00f70$@com> References: <01ac01c9e533$e0b79430$a226bc90$@com> <01e901c9e536$6d8aafd0$48a00f70$@com> Message-ID: If FreeSWITCH segfaulted its a bug we should fix it... we have no carpet so its impossible to sweet things under the rug.. please report the crash to jira. /b On Jun 4, 2009, at 12:03 PM, Ing. Edwin Villarreal wrote: > Thanks Brian. > > After commenting mod_openzap in modules.conf.xml I get to run FS > with no problem. > > Problem is with Zaptel / Wanpipe interface. > > L > > Ing. Edwin Villarreal > World Net Commerce SA CV > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3dbe7fa1/attachment-0002.html From regs at kinetix.gr Thu Jun 4 10:47:21 2009 From: regs at kinetix.gr (regs at kinetix.gr) Date: Thu, 04 Jun 2009 20:47:21 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> Message-ID: <4A2808A9.8070409@kinetix.gr> Michael Collins wrote: > > > The dialplan : > > > > > > > > expression="^.*$"> > > > You forgot the parens around .* > It should be expression="^(.*)$" if you plan to use $1 later in the > extension... > > > > > data="absolute_codec_string=PCMA"/> > data="sofia/gateway/sipp01/$1"/> > > ... like here ^^^^^^^ > :) > -MC You are right! Although, I don't think that would change the outcome of my test :) > > > > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Thu Jun 4 14:12:47 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 16:12:47 -0500 Subject: [Freeswitch-users] Interesting NAT issues In-Reply-To: <20090604094152.GA15717@jdc.jasonjgw.net> References: <20090604082921.GA838@jdc.jasonjgw.net> <20090604094152.GA15717@jdc.jasonjgw.net> Message-ID: While the o= line shouldn't matter we have fixed sofia to honor the info we give it and correctly give the right ip in that line also. So please update to 13621 and see if that corrects your issue. Thanks, Brian On Jun 4, 2009, at 4:41 AM, Jason White wrote: > Michael Jerris wrote: >> Can you please re-test with current svn trunk. we added some new nat >> busting code yesterday that may assist with this. You will need to >> specify the new > value="localnet.auto"/ >>> param in the sofia profile (see http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/sip_profiles/internal.xml?r1=13389&r2=13595 >> ) > > If I apply this to the external profile (using it on the internal > profile has > no effect), the Via headers now receive the external IP address, > i.e., the > publicly routable address provided by the ISP. > > However, the session description still has the private address in > it, which > causes the remote end to issue the following: > SIP/2.0 488 Invalid Session Description > Warning: 301 203.2.134.1 "invalid transport IP address" > > I assume (but may be wrong - full traces can be provided if > necessary) that > the problem is in the SDP that we're sending out: > o=FreeSWITCH 7801931346801551648 4196891485744912323 IN IP4 > 192.168.0.2 > > Can this be fixed up? > > Thanks. > > Jason. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b9541d22/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 14:34:04 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 21:34:04 +0000 Subject: [Freeswitch-users] WikiPBX Installation Message-ID: <4A283DCC.5040701@gmail.com> hello my friends please i need any Web FrontEnd for FS bicose i'm blind i'm using a screen reader. this screen reader is for windows only also is don't support a consol interface only graphical interface and i have a CentOs4.7 Box that contin my FS installed but i need to manage it (extentions, gateways and other fiturs please anyone setup WikiPBX for me? any help is welcome thanks my friends From matthew.lockwood at gmail.com Thu Jun 4 14:39:30 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 14:39:30 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A283DCC.5040701@gmail.com> References: <4A283DCC.5040701@gmail.com> Message-ID: <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> I'm about to start creating one. I think FS needs a UI comparable to FreePBX or similar. It's the next project on my list. It would certainly be a boost to the FS project and user acceptance. On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: > hello my friends > > please i need any Web FrontEnd for FS > bicose i'm blind i'm using a screen reader. > this screen reader is for windows only > also is don't support a consol interface only graphical interface > and i have a CentOs4.7 Box that contin my FS installed > but i need to manage it (extentions, gateways and other fiturs > please anyone setup WikiPBX for me? > any help is welcome > thanks my friends > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ba232a78/attachment-0002.html From brian at freeswitch.org Thu Jun 4 14:45:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 16:45:08 -0500 Subject: [Freeswitch-users] IPv6 registration fails under rev. 13606 In-Reply-To: <20090604062323.GA19740@jdc.jasonjgw.net> References: <20090604053416.GA9751@jdc.jasonjgw.net> <971C091E-6FEC-471A-855B-66FC976D4E2C@freeswitch.org> <20090604062323.GA19740@jdc.jasonjgw.net> Message-ID: <66DC522C-1874-44C1-9D76-8AF8536CC42A@freeswitch.org> Jason, Can I appoint you the official ipv6 tester? :) Anyway update to trunk and see if everything is ok still. Thanks, Brian On Jun 4, 2009, at 1:23 AM, Jason White wrote: > Brian West wrote: >> A few more checks went in try 13610... See I knew I would introduce >> some >> regressions. :( > > Works for me, thanks! > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/02213d61/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 14:56:13 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 21:56:13 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> Message-ID: <4A2842FD.304@gmail.com> hello, please: * in each programing language you want to start this project? * how i can contribute to it? thanks Matthew Lockwood wrote: > I'm about to start creating one. I think FS needs a UI comparable to > FreePBX or similar. It's the next project on my list. It would > certainly be a boost to the FS project and user acceptance. > > On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb > wrote: > > hello my friends > > please i need any Web FrontEnd for FS > bicose i'm blind i'm using a screen reader. > this screen reader is for windows only > also is don't support a consol interface only graphical interface > and i have a CentOs4.7 Box that contin my FS installed > but i need to manage it (extentions, gateways and other fiturs > please anyone setup WikiPBX for me? > any help is welcome > thanks my friends > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/cc214931/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:02:10 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:02:10 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A2842FD.304@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> Message-ID: <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> This may raise eyebrows, but I was thinking about using the dotnet/mono framework. I'm coming from 10+ years in Windows, and have blindly followed MS technologies through my career, so I'm very comfortable with C#. There are inherent issues with this approach though. The other alternative is PHP, although it's not my first choice and I'm nowhere near as comfortable with it. What're your thoughts? On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: > hello, > please: > > - in each programing language you want to start this project? > - how i can contribute to it? > > thanks > Matthew Lockwood wrote: > > I'm about to start creating one. I think FS needs a UI comparable to > FreePBX or similar. It's the next project on my list. It would certainly be > a boost to the FS project and user acceptance. > > On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: > >> hello my friends >> >> please i need any Web FrontEnd for FS >> bicose i'm blind i'm using a screen reader. >> this screen reader is for windows only >> also is don't support a consol interface only graphical interface >> and i have a CentOs4.7 Box that contin my FS installed >> but i need to manage it (extentions, gateways and other fiturs >> please anyone setup WikiPBX for me? >> any help is welcome >> thanks my friends >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/fb539e22/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 15:05:46 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:05:46 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> Message-ID: <4A28453A.6040205@gmail.com> hello i'm also a windows Developer / microsoft technologie user i use.Net/Mono/... #Develope, SQLite and ... are you in irc now? my nick: DelphiWorld please message me. thanks Matthew Lockwood wrote: > This may raise eyebrows, but I was thinking about using the > dotnet/mono framework. I'm coming from 10+ years in Windows, and have > blindly followed MS technologies through my career, so I'm very > comfortable with C#. There are inherent issues with this approach > though. The other alternative is PHP, although it's not my first > choice and I'm nowhere near as comfortable with it. > > What're your thoughts? > > On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb > wrote: > > hello, > please: > > * in each programing language you want to start this project? > * how i can contribute to it? > > thanks > Matthew Lockwood wrote: >> I'm about to start creating one. I think FS needs a UI comparable >> to FreePBX or similar. It's the next project on my list. It would >> certainly be a boost to the FS project and user acceptance. >> >> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >> > wrote: >> >> hello my friends >> >> please i need any Web FrontEnd for FS >> bicose i'm blind i'm using a screen reader. >> this screen reader is for windows only >> also is don't support a consol interface only graphical interface >> and i have a CentOs4.7 Box that contin my FS installed >> but i need to manage it (extentions, gateways and other fiturs >> please anyone setup WikiPBX for me? >> any help is welcome >> thanks my friends >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/86c240bf/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 4 15:08:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 4 Jun 2009 17:08:35 -0500 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A2808A9.8070409@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> <4A2808A9.8070409@kinetix.gr> Message-ID: <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> FS uses async rtp timers so you may want to set rtp-timer-name=none in the profile param to simulate asterisk conditions. Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit single cpu box because that was what was popular when it was designed and the chance for race conditions is minimal because there is only 1 cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic difference. I will be happy to investigate this issue a bit if you'd like but i do not have any box like you describe so if I can't find anything you may have to lend us your lab. On Thu, Jun 4, 2009 at 12:47 PM, regs at kinetix.gr wrote: > Michael Collins wrote: > > > > > > The dialplan : > > > > > > > > > > > > > > > > > expression="^.*$"> > > > > > > You forgot the parens around .* > > It should be expression="^(.*)$" if you plan to use $1 later in the > > extension... > > > > > > > > > > > data="absolute_codec_string=PCMA"/> > > > data="sofia/gateway/sipp01/$1"/> > > > > ... like here ^^^^^^^ > > :) > > -MC > > You are right! Although, I don't think that would change the outcome of > my test :) > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/459b5035/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:09:16 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:09:16 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A28453A.6040205@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> Message-ID: <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> KeeblerElfMatt ... I'll be there in a minute On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: > hello > i'm also a windows Developer / microsoft technologie user > i use.Net/Mono/... #Develope, SQLite and ... > are you in irc now? > my nick: DelphiWorld > please message me. > thanks > Matthew Lockwood wrote: > > This may raise eyebrows, but I was thinking about using the dotnet/mono > framework. I'm coming from 10+ years in Windows, and have blindly followed > MS technologies through my career, so I'm very comfortable with C#. There > are inherent issues with this approach though. The other alternative is PHP, > although it's not my first choice and I'm nowhere near as comfortable with > it. > > What're your thoughts? > > On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: > >> hello, >> please: >> >> - in each programing language you want to start this project? >> - how i can contribute to it? >> >> thanks >> Matthew Lockwood wrote: >> >> I'm about to start creating one. I think FS needs a UI comparable to >> FreePBX or similar. It's the next project on my list. It would certainly be >> a boost to the FS project and user acceptance. >> >> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >> >>> hello my friends >>> >>> please i need any Web FrontEnd for FS >>> bicose i'm blind i'm using a screen reader. >>> this screen reader is for windows only >>> also is don't support a consol interface only graphical interface >>> and i have a CentOs4.7 Box that contin my FS installed >>> but i need to manage it (extentions, gateways and other fiturs >>> please anyone setup WikiPBX for me? >>> any help is welcome >>> thanks my friends >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/12fb4c92/attachment-0002.html From intralanman at freeswitch.org Thu Jun 4 15:16:27 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 04 Jun 2009 18:16:27 -0400 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> Message-ID: <4A2847BB.8020306@freeswitch.org> I think you're more likely to get more help if you write it in PHP or ruby. PHP seems to be the open-source web language of choice by most, but ruby/rails has been gaining a lot of ground, so either would probably be a safe bet. -Ray Matthew Lockwood wrote: > KeeblerElfMatt ... I'll be there in a minute > > On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb > wrote: > > hello > i'm also a windows Developer / microsoft technologie user > i use.Net/Mono/... #Develope, SQLite and ... > are you in irc now? > my nick: DelphiWorld > please message me. > thanks > Matthew Lockwood wrote: >> This may raise eyebrows, but I was thinking about using the >> dotnet/mono framework. I'm coming from 10+ years in Windows, and >> have blindly followed MS technologies through my career, so I'm >> very comfortable with C#. There are inherent issues with this >> approach though. The other alternative is PHP, although it's not >> my first choice and I'm nowhere near as comfortable with it. >> >> What're your thoughts? >> >> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >> > wrote: >> >> hello, >> please: >> >> * in each programing language you want to start this project? >> * how i can contribute to it? >> >> thanks >> Matthew Lockwood wrote: >>> I'm about to start creating one. I think FS needs a UI >>> comparable to FreePBX or similar. It's the next project on >>> my list. It would certainly be a boost to the FS project and >>> user acceptance. >>> >>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>> > wrote: >>> >>> hello my friends >>> >>> please i need any Web FrontEnd for FS >>> bicose i'm blind i'm using a screen reader. >>> this screen reader is for windows only >>> also is don't support a consol interface only graphical >>> interface >>> and i have a CentOs4.7 Box that contin my FS installed >>> but i need to manage it (extentions, gateways and other >>> fiturs >>> please anyone setup WikiPBX for me? >>> any help is welcome >>> thanks my friends >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d389a666/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 15:22:20 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:22:20 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A2847BB.8020306@freeswitch.org> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> Message-ID: <4A28491C.7050009@gmail.com> hello yes, PHP/Ruby is the best Web Scripting Languages but each one in the World have there knoledge base limitation thanks Raymond Chandler wrote: > I think you're more likely to get more help if you write it in PHP or > ruby. PHP seems to be the open-source web language of choice by most, > but ruby/rails has been gaining a lot of ground, so either would > probably be a safe bet. > > -Ray > > Matthew Lockwood wrote: >> KeeblerElfMatt ... I'll be there in a minute >> >> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb > > wrote: >> >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >>> This may raise eyebrows, but I was thinking about using the >>> dotnet/mono framework. I'm coming from 10+ years in Windows, and >>> have blindly followed MS technologies through my career, so I'm >>> very comfortable with C#. There are inherent issues with this >>> approach though. The other alternative is PHP, although it's not >>> my first choice and I'm nowhere near as comfortable with it. >>> >>> What're your thoughts? >>> >>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>> > wrote: >>> >>> hello, >>> please: >>> >>> * in each programing language you want to start this >>> project? >>> * how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>>> I'm about to start creating one. I think FS needs a UI >>>> comparable to FreePBX or similar. It's the next project on >>>> my list. It would certainly be a boost to the FS project >>>> and user acceptance. >>>> >>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>> > wrote: >>>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only graphical >>>> interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and other >>>> fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ Freeswitch-users >>> mailing list Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/c771db36/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:23:01 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:23:01 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A2847BB.8020306@freeswitch.org> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> Message-ID: <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> There are two areas of help I'll need for sure: 1) Understanding certain nuances of FreeSWITCH and how to make certain things work (there are things I'll need to implement that aren't in my implementation) 2) User testing and lots and lots of feedback. I was planning to initially adopt the FreePBX look. If there's a web designer that wants to put together a new interface, that would most certainly be a welcome addition. On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler wrote: > I think you're more likely to get more help if you write it in PHP or > ruby. PHP seems to be the open-source web language of choice by most, but > ruby/rails has been gaining a lot of ground, so either would probably be a > safe bet. > > -Ray > > Matthew Lockwood wrote: > > KeeblerElfMatt ... I'll be there in a minute > > On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: > >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >> >> This may raise eyebrows, but I was thinking about using the dotnet/mono >> framework. I'm coming from 10+ years in Windows, and have blindly followed >> MS technologies through my career, so I'm very comfortable with C#. There >> are inherent issues with this approach though. The other alternative is PHP, >> although it's not my first choice and I'm nowhere near as comfortable with >> it. >> >> What're your thoughts? >> >> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >> >>> hello, >>> please: >>> >>> - in each programing language you want to start this project? >>> - how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>> >>> I'm about to start creating one. I think FS needs a UI comparable to >>> FreePBX or similar. It's the next project on my list. It would certainly be >>> a boost to the FS project and user acceptance. >>> >>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only graphical interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and other fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/41025f7f/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:25:03 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:25:03 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A28491C.7050009@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <4A28491C.7050009@gmail.com> Message-ID: <415541b10906041525l25b6cf3ak56d390e113b0fc9d@mail.gmail.com> Ray - The other issue is my competence in those scripting languages; I'm very competent in C# and the .NET framework. I'll have a substantial amount of downtime while I'm on several long-haul flights in the near future, so I can devote a good amount of time to getting this done. Mefta - I'm in IRC right now. On Thu, Jun 4, 2009 at 3:22 PM, Meftah Tayeb wrote: > hello > yes, PHP/Ruby is the best Web Scripting Languages > but each one in the World have there knoledge base limitation > thanks > Raymond Chandler wrote: > > I think you're more likely to get more help if you write it in PHP or ruby. > PHP seems to be the open-source web language of choice by most, but > ruby/rails has been gaining a lot of ground, so either would probably be a > safe bet. > > -Ray > > Matthew Lockwood wrote: > > KeeblerElfMatt ... I'll be there in a minute > > On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: > >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >> >> This may raise eyebrows, but I was thinking about using the dotnet/mono >> framework. I'm coming from 10+ years in Windows, and have blindly followed >> MS technologies through my career, so I'm very comfortable with C#. There >> are inherent issues with this approach though. The other alternative is PHP, >> although it's not my first choice and I'm nowhere near as comfortable with >> it. >> >> What're your thoughts? >> >> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >> >>> hello, >>> please: >>> >>> - in each programing language you want to start this project? >>> - how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>> >>> I'm about to start creating one. I think FS needs a UI comparable to >>> FreePBX or similar. It's the next project on my list. It would certainly be >>> a boost to the FS project and user acceptance. >>> >>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only graphical interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and other fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/12292170/attachment-0002.html From brian at freeswitch.org Thu Jun 4 15:27:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 17:27:55 -0500 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: ACK!~ :P Don't copy, be original... thats how you win! /b On Jun 4, 2009, at 5:23 PM, Matthew Lockwood wrote: > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ecb91011/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 15:28:32 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:28:32 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: <4A284A90.70602@gmail.com> hi, yes, i'm here to help you developing this Web GUI bicose only WikiPBX is here now and is not easy to install i hop that you start one with me thanks Matthew Lockwood wrote: > There are two areas of help I'll need for sure: > 1) Understanding certain nuances of FreeSWITCH and how to make certain > things work (there are things I'll need to implement that aren't in my > implementation) > 2) User testing and lots and lots of feedback. > > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. > > On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler > > wrote: > > I think you're more likely to get more help if you write it in PHP > or ruby. PHP seems to be the open-source web language of choice by > most, but ruby/rails has been gaining a lot of ground, so either > would probably be a safe bet. > > -Ray > > Matthew Lockwood wrote: >> KeeblerElfMatt ... I'll be there in a minute >> >> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb >> > wrote: >> >> hello >> i'm also a windows Developer / microsoft technologie user >> i use.Net/Mono/... #Develope, SQLite and ... >> are you in irc now? >> my nick: DelphiWorld >> please message me. >> thanks >> Matthew Lockwood wrote: >>> This may raise eyebrows, but I was thinking about using the >>> dotnet/mono framework. I'm coming from 10+ years in Windows, >>> and have blindly followed MS technologies through my career, >>> so I'm very comfortable with C#. There are inherent issues >>> with this approach though. The other alternative is PHP, >>> although it's not my first choice and I'm nowhere near as >>> comfortable with it. >>> >>> What're your thoughts? >>> >>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>> > wrote: >>> >>> hello, >>> please: >>> >>> * in each programing language you want to start this >>> project? >>> * how i can contribute to it? >>> >>> thanks >>> Matthew Lockwood wrote: >>>> I'm about to start creating one. I think FS needs a UI >>>> comparable to FreePBX or similar. It's the next project >>>> on my list. It would certainly be a boost to the FS >>>> project and user acceptance. >>>> >>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>> >>> > wrote: >>>> >>>> hello my friends >>>> >>>> please i need any Web FrontEnd for FS >>>> bicose i'm blind i'm using a screen reader. >>>> this screen reader is for windows only >>>> also is don't support a consol interface only >>>> graphical interface >>>> and i have a CentOs4.7 Box that contin my FS installed >>>> but i need to manage it (extentions, gateways and >>>> other fiturs >>>> please anyone setup WikiPBX for me? >>>> any help is welcome >>>> thanks my friends >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/de20205a/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 15:30:38 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:30:38 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041525l25b6cf3ak56d390e113b0fc9d@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <4A28491C.7050009@gmail.com> <415541b10906041525l25b6cf3ak56d390e113b0fc9d@mail.gmail.com> Message-ID: <4A284B0E.6000900@gmail.com> hello please send to me a private Message (DelphiWorld) i can't find your Nick name thanks Matthew Lockwood wrote: > Ray - The other issue is my competence in those scripting languages; > I'm very competent in C# and the .NET framework. I'll have a > substantial amount of downtime while I'm on several long-haul flights > in the near future, so I can devote a good amount of time to getting > this done. > > Mefta - I'm in IRC right now. > > On Thu, Jun 4, 2009 at 3:22 PM, Meftah Tayeb > wrote: > > hello > yes, PHP/Ruby is the best Web Scripting Languages > but each one in the World have there knoledge base limitation > thanks > Raymond Chandler wrote: >> I think you're more likely to get more help if you write it in >> PHP or ruby. PHP seems to be the open-source web language of >> choice by most, but ruby/rails has been gaining a lot of ground, >> so either would probably be a safe bet. >> >> -Ray >> >> Matthew Lockwood wrote: >>> KeeblerElfMatt ... I'll be there in a minute >>> >>> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb >>> > wrote: >>> >>> hello >>> i'm also a windows Developer / microsoft technologie user >>> i use.Net/Mono/... #Develope, SQLite and ... >>> are you in irc now? >>> my nick: DelphiWorld >>> please message me. >>> thanks >>> Matthew Lockwood wrote: >>>> This may raise eyebrows, but I was thinking about using the >>>> dotnet/mono framework. I'm coming from 10+ years in >>>> Windows, and have blindly followed MS technologies through >>>> my career, so I'm very comfortable with C#. There are >>>> inherent issues with this approach though. The other >>>> alternative is PHP, although it's not my first choice and >>>> I'm nowhere near as comfortable with it. >>>> >>>> What're your thoughts? >>>> >>>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>>> > wrote: >>>> >>>> hello, >>>> please: >>>> >>>> * in each programing language you want to start >>>> this project? >>>> * how i can contribute to it? >>>> >>>> thanks >>>> Matthew Lockwood wrote: >>>>> I'm about to start creating one. I think FS needs a UI >>>>> comparable to FreePBX or similar. It's the next >>>>> project on my list. It would certainly be a boost to >>>>> the FS project and user acceptance. >>>>> >>>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>>> >>>> > wrote: >>>>> >>>>> hello my friends >>>>> >>>>> please i need any Web FrontEnd for FS >>>>> bicose i'm blind i'm using a screen reader. >>>>> this screen reader is for windows only >>>>> also is don't support a consol interface only >>>>> graphical interface >>>>> and i have a CentOs4.7 Box that contin my FS installed >>>>> but i need to manage it (extentions, gateways and >>>>> other fiturs >>>>> please anyone setup WikiPBX for me? >>>>> any help is welcome >>>>> thanks my friends >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ac11fec7/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 15:31:41 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:31:41 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: <4A284B4D.5000004@gmail.com> hello, .Net is no only for windows but Linux also using MONO and #develope is open source thanks Brian West wrote: > ACK!~ :P Don't copy, be original... thats how you win! > > /b > > On Jun 4, 2009, at 5:23 PM, Matthew Lockwood wrote: > >> I was planning to initially adopt the FreePBX look. If there's a web >> designer that wants to put together a new interface, that would most >> certainly be a welcome addition. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/24dd9ecb/attachment-0002.html From msc at freeswitch.org Thu Jun 4 15:34:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 15:34:32 -0700 Subject: [Freeswitch-users] ATTENTION: Outstanding JIRA Issues Message-ID: <87f2f3b90906041534y316eda97m96c98883a1f1ed7b@mail.gmail.com> To all FreeSWITCHers, If you have any open issues on JIRA please tend to them ASAP. If you have any issues that are not yet reported please report them ASAP. If you have bugs you are not sure about please join IRC and ask for assistance. We are pushing very hard to get as many bugs resolved as possible prior to 1.0.4. Thanks for your assistance with making FreeSWITCH such an awesome project with a great community! -Michael http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/815f955f/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:34:43 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:34:43 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A284A90.70602@gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> Message-ID: <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> An slightly more graphical install wizard would actually be useful too. This could be extended into something that could install and configure FreeSWITCH and install the GUI all at the same time. Talk about lowering the barrier to entry! Brian - you're completely right. This idea has only been floating around in my head for 48 hours so it's not very mature. As we all know, it's 80% marketing, and 20% product (no smack talk, but think Asterisk!). I'll come up with a strategy. On Thu, Jun 4, 2009 at 3:28 PM, Meftah Tayeb wrote: > hi, > yes, i'm here to help you developing this Web GUI > bicose only WikiPBX is here now and is not easy to install > i hop that you start one with me > thanks > Matthew Lockwood wrote: > > There are two areas of help I'll need for sure: > 1) Understanding certain nuances of FreeSWITCH and how to make certain > things work (there are things I'll need to implement that aren't in my > implementation) > 2) User testing and lots and lots of feedback. > > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. > > On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler < > intralanman at freeswitch.org> wrote: > >> I think you're more likely to get more help if you write it in PHP or >> ruby. PHP seems to be the open-source web language of choice by most, but >> ruby/rails has been gaining a lot of ground, so either would probably be a >> safe bet. >> >> -Ray >> >> Matthew Lockwood wrote: >> >> KeeblerElfMatt ... I'll be there in a minute >> >> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: >> >>> hello >>> i'm also a windows Developer / microsoft technologie user >>> i use.Net/Mono/... #Develope, SQLite and ... >>> are you in irc now? >>> my nick: DelphiWorld >>> please message me. >>> thanks >>> Matthew Lockwood wrote: >>> >>> This may raise eyebrows, but I was thinking about using the dotnet/mono >>> framework. I'm coming from 10+ years in Windows, and have blindly followed >>> MS technologies through my career, so I'm very comfortable with C#. There >>> are inherent issues with this approach though. The other alternative is PHP, >>> although it's not my first choice and I'm nowhere near as comfortable with >>> it. >>> >>> What're your thoughts? >>> >>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >>> >>>> hello, >>>> please: >>>> >>>> - in each programing language you want to start this project? >>>> - how i can contribute to it? >>>> >>>> thanks >>>> Matthew Lockwood wrote: >>>> >>>> I'm about to start creating one. I think FS needs a UI comparable to >>>> FreePBX or similar. It's the next project on my list. It would certainly be >>>> a boost to the FS project and user acceptance. >>>> >>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>>> >>>>> hello my friends >>>>> >>>>> please i need any Web FrontEnd for FS >>>>> bicose i'm blind i'm using a screen reader. >>>>> this screen reader is for windows only >>>>> also is don't support a consol interface only graphical interface >>>>> and i have a CentOs4.7 Box that contin my FS installed >>>>> but i need to manage it (extentions, gateways and other fiturs >>>>> please anyone setup WikiPBX for me? >>>>> any help is welcome >>>>> thanks my friends >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/548a5d37/attachment-0002.html From tayeb.meftah at gmail.com Thu Jun 4 15:38:05 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 04 Jun 2009 22:38:05 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> Message-ID: <4A284CCD.6030806@gmail.com> hellok, do you need a graphical installer? i'm here to create it for the community! please tel me how i start a new FS subproject thanks Matthew Lockwood wrote: > An slightly more graphical install wizard would actually be useful > too. This could be extended into something that could install and > configure FreeSWITCH and install the GUI all at the same time. Talk > about lowering the barrier to entry! > > Brian - you're completely right. This idea has only been floating > around in my head for 48 hours so it's not very mature. As we all > know, it's 80% marketing, and 20% product (no smack talk, but think > Asterisk!). I'll come up with a strategy. > > On Thu, Jun 4, 2009 at 3:28 PM, Meftah Tayeb > wrote: > > hi, > yes, i'm here to help you developing this Web GUI > bicose only WikiPBX is here now and is not easy to install > i hop that you start one with me > thanks > Matthew Lockwood wrote: >> There are two areas of help I'll need for sure: >> 1) Understanding certain nuances of FreeSWITCH and how to make >> certain things work (there are things I'll need to implement that >> aren't in my implementation) >> 2) User testing and lots and lots of feedback. >> >> I was planning to initially adopt the FreePBX look. If there's a >> web designer that wants to put together a new interface, that >> would most certainly be a welcome addition. >> >> On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler >> > >> wrote: >> >> I think you're more likely to get more help if you write it >> in PHP or ruby. PHP seems to be the open-source web language >> of choice by most, but ruby/rails has been gaining a lot of >> ground, so either would probably be a safe bet. >> >> -Ray >> >> Matthew Lockwood wrote: >>> KeeblerElfMatt ... I'll be there in a minute >>> >>> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb >>> > wrote: >>> >>> hello >>> i'm also a windows Developer / microsoft technologie user >>> i use.Net/Mono/... #Develope, SQLite and ... >>> are you in irc now? >>> my nick: DelphiWorld >>> please message me. >>> thanks >>> Matthew Lockwood wrote: >>>> This may raise eyebrows, but I was thinking about using >>>> the dotnet/mono framework. I'm coming from 10+ years in >>>> Windows, and have blindly followed MS technologies >>>> through my career, so I'm very comfortable with C#. >>>> There are inherent issues with this approach though. >>>> The other alternative is PHP, although it's not my >>>> first choice and I'm nowhere near as comfortable with it. >>>> >>>> What're your thoughts? >>>> >>>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb >>>> >>> > wrote: >>>> >>>> hello, >>>> please: >>>> >>>> * in each programing language you want to start >>>> this project? >>>> * how i can contribute to it? >>>> >>>> thanks >>>> Matthew Lockwood wrote: >>>>> I'm about to start creating one. I think FS needs >>>>> a UI comparable to FreePBX or similar. It's the >>>>> next project on my list. It would certainly be a >>>>> boost to the FS project and user acceptance. >>>>> >>>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb >>>>> >>>> > wrote: >>>>> >>>>> hello my friends >>>>> >>>>> please i need any Web FrontEnd for FS >>>>> bicose i'm blind i'm using a screen reader. >>>>> this screen reader is for windows only >>>>> also is don't support a consol interface only >>>>> graphical interface >>>>> and i have a CentOs4.7 Box that contin my FS >>>>> installed >>>>> but i need to manage it (extentions, gateways >>>>> and other fiturs >>>>> please anyone setup WikiPBX for me? >>>>> any help is welcome >>>>> thanks my friends >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/d65b467d/attachment-0002.html From msc at freeswitch.org Thu Jun 4 15:39:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 15:39:13 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> Message-ID: <87f2f3b90906041539r47bdf2c6m1df4671ad4f67342@mail.gmail.com> On Thu, Jun 4, 2009 at 2:39 PM, Matthew Lockwood wrote: > I'm about to start creating one. I think FS needs a UI comparable to > FreePBX or similar. It's the next project on my list. It would certainly be > a boost to the FS project and user acceptance. > Be sure to check with Bougyman on IRC. He is planning on releasing an open-source, MIT-licensed FS-GUI w/ underlying framework. Last I heard he said week of June 15. Note: he said that it will require Ruby plus Rack and PostgreSQL. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/c11fa02e/attachment-0002.html From msc at freeswitch.org Thu Jun 4 15:40:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 15:40:10 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> Message-ID: <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> On Thu, Jun 4, 2009 at 3:34 PM, Matthew Lockwood wrote: > An slightly more graphical install wizard would actually be useful too. > This could be extended into something that could install and configure > FreeSWITCH and install the GUI all at the same time. Talk about lowering the > barrier to entry! > > Brian - you're completely right. This idea has only been floating around in > my head for 48 hours so it's not very mature. As we all know, it's 80% > marketing, and 20% product (no smack talk, but think Asterisk!). I'll come > up with a strategy. well, the numbers are more like 98.5% marketing in that case. :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b3084ccb/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:41:32 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:41:32 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <4A284CCD.6030806@gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <4A284CCD.6030806@gmail.com> Message-ID: <415541b10906041541s3f01355axef05b978a7c1f5bf@mail.gmail.com> That would be incredible of you. Let's lower the barrier to entry to total noobs! The simpler we can make it the more successful the project will be. On Thu, Jun 4, 2009 at 3:38 PM, Meftah Tayeb wrote: > hellok, > do you need a graphical installer? > i'm here to create it for the community! > please tel me how i start a new FS subproject > thanks > Matthew Lockwood wrote: > > An slightly more graphical install wizard would actually be useful too. > This could be extended into something that could install and configure > FreeSWITCH and install the GUI all at the same time. Talk about lowering the > barrier to entry! > > Brian - you're completely right. This idea has only been floating around in > my head for 48 hours so it's not very mature. As we all know, it's 80% > marketing, and 20% product (no smack talk, but think Asterisk!). I'll come > up with a strategy. > > On Thu, Jun 4, 2009 at 3:28 PM, Meftah Tayeb wrote: > >> hi, >> yes, i'm here to help you developing this Web GUI >> bicose only WikiPBX is here now and is not easy to install >> i hop that you start one with me >> thanks >> Matthew Lockwood wrote: >> >> There are two areas of help I'll need for sure: >> 1) Understanding certain nuances of FreeSWITCH and how to make certain >> things work (there are things I'll need to implement that aren't in my >> implementation) >> 2) User testing and lots and lots of feedback. >> >> I was planning to initially adopt the FreePBX look. If there's a web >> designer that wants to put together a new interface, that would most >> certainly be a welcome addition. >> >> On Thu, Jun 4, 2009 at 3:16 PM, Raymond Chandler < >> intralanman at freeswitch.org> wrote: >> >>> I think you're more likely to get more help if you write it in PHP or >>> ruby. PHP seems to be the open-source web language of choice by most, but >>> ruby/rails has been gaining a lot of ground, so either would probably be a >>> safe bet. >>> >>> -Ray >>> >>> Matthew Lockwood wrote: >>> >>> KeeblerElfMatt ... I'll be there in a minute >>> >>> On Thu, Jun 4, 2009 at 3:05 PM, Meftah Tayeb wrote: >>> >>>> hello >>>> i'm also a windows Developer / microsoft technologie user >>>> i use.Net/Mono/... #Develope, SQLite and ... >>>> are you in irc now? >>>> my nick: DelphiWorld >>>> please message me. >>>> thanks >>>> Matthew Lockwood wrote: >>>> >>>> This may raise eyebrows, but I was thinking about using the >>>> dotnet/mono framework. I'm coming from 10+ years in Windows, and have >>>> blindly followed MS technologies through my career, so I'm very comfortable >>>> with C#. There are inherent issues with this approach though. The other >>>> alternative is PHP, although it's not my first choice and I'm nowhere near >>>> as comfortable with it. >>>> >>>> What're your thoughts? >>>> >>>> On Thu, Jun 4, 2009 at 2:56 PM, Meftah Tayeb wrote: >>>> >>>>> hello, >>>>> please: >>>>> >>>>> - in each programing language you want to start this project? >>>>> - how i can contribute to it? >>>>> >>>>> thanks >>>>> Matthew Lockwood wrote: >>>>> >>>>> I'm about to start creating one. I think FS needs a UI comparable to >>>>> FreePBX or similar. It's the next project on my list. It would certainly be >>>>> a boost to the FS project and user acceptance. >>>>> >>>>> On Thu, Jun 4, 2009 at 2:34 PM, Meftah Tayeb wrote: >>>>> >>>>>> hello my friends >>>>>> >>>>>> please i need any Web FrontEnd for FS >>>>>> bicose i'm blind i'm using a screen reader. >>>>>> this screen reader is for windows only >>>>>> also is don't support a consol interface only graphical interface >>>>>> and i have a CentOs4.7 Box that contin my FS installed >>>>>> but i need to manage it (extentions, gateways and other fiturs >>>>>> please anyone setup WikiPBX for me? >>>>>> any help is welcome >>>>>> thanks my friends >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/521658d2/attachment-0002.html From brian at freeswitch.org Thu Jun 4 15:41:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 17:41:55 -0500 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> Message-ID: <90DA2AF2-1BDF-4FBE-8732-626B31874B6F@freeswitch.org> Its called marchitecture! /b On Jun 4, 2009, at 5:40 PM, Michael Collins wrote: > well, the numbers are more like 98.5% marketing in that case. :) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From matthew.lockwood at gmail.com Thu Jun 4 15:42:48 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:42:48 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> Message-ID: <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> LOL. I'd rather have a mediocre product and excellent marketing than the other way around; building a better mouse trap and sitting back is totally ineffective. A second GUI wouldn't be a bad thing. On Thu, Jun 4, 2009 at 3:40 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 3:34 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> An slightly more graphical install wizard would actually be useful too. >> This could be extended into something that could install and configure >> FreeSWITCH and install the GUI all at the same time. Talk about lowering the >> barrier to entry! >> >> Brian - you're completely right. This idea has only been floating around >> in my head for 48 hours so it's not very mature. As we all know, it's 80% >> marketing, and 20% product (no smack talk, but think Asterisk!). I'll come >> up with a strategy. > > > well, the numbers are more like 98.5% marketing in that case. :) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/ef0e66a1/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:44:34 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:44:34 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> Message-ID: <415541b10906041544p349b408dg3073ec5cb771194b@mail.gmail.com> You're very, very right. Okay, so I'm going to need a web designer in the community to work with! Given that this idea has only been floating around in my mind for 48 hours, I've barely thought about it. Although this is a FOSS project, I should apply all the rules of business I've been successfully applying for years. On Thu, Jun 4, 2009 at 3:27 PM, Brian West wrote: > ACK!~ :P Don't copy, be original... thats how you win! > /b > > On Jun 4, 2009, at 5:23 PM, Matthew Lockwood wrote: > > I was planning to initially adopt the FreePBX look. If there's a web > designer that wants to put together a new interface, that would most > certainly be a welcome addition. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/5e891335/attachment-0002.html From brian at freeswitch.org Thu Jun 4 15:46:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 17:46:37 -0500 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2842FD.304@gmail.com> <415541b10906041502r1d8339dap99bb08fe31ef07ff@mail.gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> Message-ID: <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> I could agree the community needs to come together and JUST DO IT... but I can't count the times I have seen this thread before. /b On Jun 4, 2009, at 5:42 PM, Matthew Lockwood wrote: > LOL. I'd rather have a mediocre product and excellent marketing than > the other way around; building a better mouse trap and sitting back > is totally ineffective. > > A second GUI wouldn't be a bad thing. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0fcb2889/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 15:48:09 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 15:48:09 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> References: <4A283DCC.5040701@gmail.com> <4A28453A.6040205@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> Message-ID: <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> Okay, I'm going to schedule in a start date of the 16th. This will have my company backing it so it will be done! On Thu, Jun 4, 2009 at 3:46 PM, Brian West wrote: > I could agree the community needs to come together and JUST DO IT... but I > can't count the times I have seen this thread before. > /b > > On Jun 4, 2009, at 5:42 PM, Matthew Lockwood wrote: > > LOL. I'd rather have a mediocre product and excellent marketing than the > other way around; building a better mouse trap and sitting back is totally > ineffective. > > A second GUI wouldn't be a bad thing. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/3a7d7d36/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 16:13:53 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 16:13:53 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041509n5f7b4a13r363a5bed987dce94@mail.gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> Message-ID: <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> Okay, leave this with me - I'll bring this together and do what's required. I'll see if I can squeeze it in ahead of schedule, but don't count on it. Also expect pleas for help at some stage, and I'll need a UI developer to create an awesome interface. You'll all be hearing from me shortly! M On Thu, Jun 4, 2009 at 3:48 PM, Matthew Lockwood wrote: > Okay, I'm going to schedule in a start date of the 16th. This will have my > company backing it so it will be done! > > On Thu, Jun 4, 2009 at 3:46 PM, Brian West wrote: > >> I could agree the community needs to come together and JUST DO IT... but I >> can't count the times I have seen this thread before. >> /b >> >> On Jun 4, 2009, at 5:42 PM, Matthew Lockwood wrote: >> >> LOL. I'd rather have a mediocre product and excellent marketing than the >> other way around; building a better mouse trap and sitting back is totally >> ineffective. >> >> A second GUI wouldn't be a bad thing. >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/b6cbc6a2/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 16:15:08 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 09:15:08 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> Message-ID: Hi Anthony, Traces as requested. Let me know if you want a jira opened or any further data. Regards, Jim On Thu, Jun 4, 2009 at 10:59 PM, Anthony Minessale wrote: > you should have also turned in the sip trace > sofia profile internal siptrace on > > > On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke wrote: >> >> Hi All, >> >> Looking for some debugging tips and comments on what might be causing >> the media port in the 200OK ( Answer message) to be set to 0 by >> freeswitch. ?Essentially it looks like data might be getting trampled >> somehow. >> >> Portion of 200OK going into Freeswitch >> m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 >> NSE/8000..a= >> ?fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >> 0-15..a=ptime:20..a=sendrecv.. >> >> Portion of 200OK coming out of Freeswitch >> m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100 >> NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event >> ?/8000..a=fmtp:101 0-15.. >> >> Note the media port has been set to 0 and the rtpmap for G729 is also >> not correct. ?On receipt of this bad 200Ok the originator sends a BYE. >> >> We are using FS as a B2BUA with bypass_media set to true. ?Thus IMHO >> Freeswitch should not be touching the SDP portion of the message and >> just passing it through. >> >> This can reproduce this at will, so I can collect whatever data is >> nessicary. ?I have added the sofia debug from the console. >> >> Thanks, >> -- >> Jim Burke >> Director Evolutiontel. >> http://www.evolutiontel.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net -------------- next part -------------- login as: root root at 202.xxx.xx.xx's password: Last login: Thu Jun 4 21:56:36 2009 from 60-241-91-137.static.tpgi.com.au [root at sip01 ~]# cd /usr/local/freeswitch/bin [root at sip01 bin]# ./fs_cli _____ ____ ____ _ ___ | ___/ ___| / ___| | |_ _| | |_ \___ \ | | | | | | | _| ___) | | |___| |___ | | |_| |____/ \____|_____|___| ***************************************************** * Anthony Minessale II, Ken Rice, Michael Jerris * * FreeSWITCH (http://www.freeswitch.org) * * Brought to you by ClueCon http://www.cluecon.com/ * ***************************************************** Type /help to see a list of commands +OK log level [7] freeswitch at internal> sofia USAGE: -------------------------------------------------------------------------------- sofia help sofia profile [[start|stop|restart|rescan] [reloadxml]|flush_inbound_reg [] [reboot]|[register|unregister] [|all]|killgw |[stun-auto-disable|stun-enabled] [true|false]]|siptrace [on|off] sofia status profile [ reg ] | [ pres ] sofia status gateway sofia loglevel [0-9] -------------------------------------------------------------------------------- freeswitch at internal> sofia loglevel all 9 Sofia log level for component [all] has been set to [9] freeswitch at internal> sofia profile internal siptrace on Enabled sip debugging on internal nua: nua_set_params: entering freeswitch at internal> nua((nil)): sent signal r_set_params nua((nil)): recv signal r_set_params nua: nua_stack_set_params: entering soa_set_params(static::0x9baf508, ...) called nua((nil)): event r_set_params 200 OK nua: nua_application_event: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9bcba58 from (udp/192.168.0.2:5070) has 1299 bytes, veclen = 1 recv 1299 bytes from udp/[192.168.0.2]:5060 at 23:06:35.605391: ------------------------------------------------------------------------ INVITE sip:0631000001 at 192.168.0.2:5060 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Max-Forwards: 69 Contact: To: From: "0445674567";tag=50f70f41-co453-INS001 Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE Content-Type: application/sdp Date: Thu, 04 Jun 2009 23:06:20 GMT Supported: 100rel User-Agent: ENSR2.5.4 Content-Length: 288 P-Incoming-GW: Yes P-Incoming-GW: Yes X-Max-TIMER: 7200 X-Source_IP: 202.xx.xxx.xx P-Asserted-Identity: v=0 o=- 1358368577 1358368577 IN IP4 202.xx.xxx.xx s=ENSResip c=IN IP4 202..xxx.xx.xx t=0 0 m=audio 12580 RTP/AVP 18 8 0 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=sendrecv ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9bcba58 (1299 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received INVITE sip:0631000001 at 192.168.0.2:5060 SIP/2.0 (CSeq 45301) nta: INVITE (45301) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9ccc130) called soa_set_params(static::0x9c1d4e8, ...) called nta_leg_tcreate(0x9c41a68) soa_init_offer_answer(static::0x9c1d4e8) called soa_set_remote_sdp(static::0x9c1d4e8, (nil), 0x9c85df3, 288) called nua(0x9ccc130): adding session usage tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 698 bytes of 698 to udp/192.168.0.2:5060 tport_vsend returned 698 send 698 bytes to udp/[192.168.0.2]:5060 at 23:06:35.624664: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Record-Route: Record-Route: From: "0445674567";tag=50f70f41-co453-INS001 To: Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE User-Agent: Evolutiontel SIP Service Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (45301) nua(0x9ccc130): event i_invite 100 Trying nua(0x9ccc130): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x9c1d4e8, [0xb77b7bac], [0xb77b7ba8], [(nil)]) called nua(0x9ccc130): event i_state 100 Trying nua: nua_application_event: entering 2009-06-05 09:06:35 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/0445674567 at sip.evolutiontel.net [e57f4f54-7b45-48cc-98d3-8b594bdc19e8] nua: nua_handle_bind: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [received][100] 2009-06-05 09:06:35 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1358368577 1358368577 IN IP4 202.xx.xxx.xx s=ENSResip c=IN IP4 202..xxx.xx.xx t=0 0 m=audio 12580 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - 2009-06-05 09:06:35 [DEBUG] sofia.c:3182 sofia_handle_sip_i_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_NEW -> CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State INIT 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA INIT 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_INIT -> CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State INIT going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State ROUTING 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/0445674567 at sip.evolutiontel.net Standard ROUTING 2009-06-05 09:06:35 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 0445674567->0631000001 in context evolutiontel Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unloop] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->global] continue=true Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${network_addr}(192.168.0.2) =~ /^$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net ANTI-Action set(use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [global] ${numbering_plan}() =~ /^$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set_user(default@${domain_name}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_h_X-ZRTP-On}() =~ /^Y$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net ANTI-Action set(bypass_media=true) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_secure_media}() =~ /^true$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [global] ${sip_user_agent}(ENSR2.5.4) =~ /^PolycomSound(Point|Station)IP-S(S|P)IP_\d{3,4}-UA\/((3).(\d).(\d).(\d{4}))$/ break=never Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Absolute Condition [global] Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->vmain] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [vmain] destination_number(0631000001) =~ /^121/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->vmain1] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [vmain1] destination_number(0631000001) =~ /^123/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->vmain2] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [vmain2] destination_number(0631000001) =~ /^122/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->ivr_demo] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [ivr_demo] destination_number(0631000001) =~ /^5000$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(0631000001) =~ /^5900$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(0631000001) =~ /^5901$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(0631000001) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(0631000001) =~ /^parking$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->park] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [park] destination_number(0631000001) =~ /callpark/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->unpark] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [unpark] destination_number(0631000001) =~ /pickup/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->wait] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [wait] destination_number(0631000001) =~ /^wait$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->National_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [National_calls] destination_number(0631000001) =~ /^0(2|3|4|5|7|8|9)[0-9]{8}$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->Special_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [Special_calls] destination_number(0631000001) =~ /^1[3|8][0-9]+$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->International_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (FAIL) [International_calls] destination_number(0631000001) =~ /^0011[0-9]+$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net parsing [evolutiontel->On-Net_calls] continue=false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Regex (PASS) [On-Net_calls] destination_number(0631000001) =~ /^063[0-9]{7}$/ break=on-false Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(execute_on_answer=sched_hangup +${sip_h_x-max-timer} ALLOTED_TIMEOUT) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(sip_cid_type=pid) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(continue_on_fail=79) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(bypass_media=false) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action set(ringback=%(400,200,401,450);%(400,2200,400,450)) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action pre_answer() Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action export(sip_secure_media=true) Dialplan: sofia/internal/0445674567 at sip.evolutiontel.net Action bridge({sip_from_uri=sip:${sip_from_uri}}sofia/internal/${sip_req_user}@192.168.0.2^${sip_to_uri}) 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_ROUTING -> CS_EXECUTE 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State ROUTING going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_EXECUTE 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State EXECUTE 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA EXECUTE 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/0445674567 at sip.evolutiontel.net Standard EXECUTE EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(use_profile=nat) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [use_profile]=[nat] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set_user(default at 192.168.0.2) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(bypass_media=true) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(bypass_media=true) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [bypass_media]=[true] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net hash(insert/192.168.0.2-spymap/0445674567/e57f4f54-7b45-48cc-98d3-8b594bdc19e8) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/0445674567/0631000001) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net hash(insert/192.168.0.2-last_dial/global/e57f4f54-7b45-48cc-98d3-8b594bdc19e8) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(execute_on_answer=sched_hangup +7200 ALLOTED_TIMEOUT) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [execute_on_answer]=[sched_hangup +7200 ALLOTED_TIMEOUT] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(sip_cid_type=pid) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [sip_cid_type]=[pid] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net set(continue_on_fail=79) 2009-06-05 09:06:35 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0445674567 at sip.evolutiontel.net SET [continue_on_fail]=[79] EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net bridge({sip_from_uri=sip:0445674567 at sip.evolutiontel.net}sofia/internal/0631000001 at 192.168.0.2^0312341234 at 202.xxx.xx.xx:5060) 2009-06-05 09:06:35 [DEBUG] switch_ivr_originate.c:1017 switch_ivr_originate() variable string 0 = [sip_from_uri=sip:0445674567 at sip.evolutiontel.net] 2009-06-05 09:06:35 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/0631000001 at 192.168.0.2 [d360e71f-64bd-4eb0-b99d-56af3c9e01f0] 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:2719 sofia_outgoing_channel() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_NEW -> CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_INIT 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State INIT 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/0631000001 at 192.168.0.2 SOFIA INIT nua: nh_create_handle: entering nua: nua_handle_bind: entering nua: nua_invite: entering nua(0x9cf7ed0): recv signal r_invite nua: nua_stack_set_params: entering soa_clone(static::0x9baf508, 0x9b9a050, 0x9cf7ed0) called soa_set_params(static::0x9d15ce0, ...) called soa_set_params(static::0x9d15ce0, ...) called soa_set_user_sdp(static::0x9d15ce0, (nil), 0x9cd16f7, -1) called soa_set_capability_sdp(static::0x9d15ce0, (nil), 0x9cd16f7, -1) called nua(0x9cf7ed0): adding session usage nta_leg_tcreate(0x9db94b0) soa_init_offer_answer(static::0x9d15ce0) called soa_generate_offer(static::0x9d15ce0, 0) called soa_static_offer_answer_action(0x9d15ce0, soa_generate_offer): called soa_static(0x9d15ce0, soa_generate_offer): generating local description soa_static(0x9d15ce0, soa_generate_offer): upgrade with local description soa_sdp_mode_set(0xb77b7ef4, (nil), ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.xxx.xx.xx (a local address) soa_static(0x9d15ce0, soa_generate_offer): storing local description soa_get_local_sdp(static::0x9d15ce0, [(nil)], [0xb77b7f7c], [0xb77b7f78]) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 1233 bytes of 1233 to udp/192.168.0.2:5060 tport_vsend returned 1233 send 1233 bytes to udp/[192.168.0.2]:5060 at 23:06:35.639526: ------------------------------------------------------------------------ INVITE sip:0631000001 at 192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5070;rport;branch=z9hG4bKZa7eS5j4r9a8F Max-Forwards: 68 From: "0445674567" ;tag=vQ8mytQ2mXNme To: Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Contact: User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 294 P-Incoming-GW: Yes P-Incoming-GW-1: Yes X-Max-TIMER: 7200 X-Source_IP: 202.xx.xxx.xx P-Asserted-Identity: "0445674567" v=0 o=- 7279717662006744601 8597592345307020033 IN IP4 202.xxx.xx.xx s=ENSResip c=IN IP4 202..xxx.xx.xx t=0 0 m=audio 12580 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - ------------------------------------------------------------------------ nta: sent INVITE (115960061) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9c63a58 for udp/192.168.0.2:5070 (already 0) nua(0x9cf7ed0): call state changed: init -> calling, sent offer soa_get_local_sdp(static::0x9d15ce0, [0xb77b7fa4], [0xb77b7fa0], [(nil)]) called nua(0x9cf7ed0): event i_state INVITE sent nua: nua_application_event: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c633c8 from (udp/192.168.0.2:5070) has 303 bytes, veclen = 1 recv 303 bytes from udp/[192.168.0.2]:5060 at 23:06:35.640683: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2:5070;rport=5070;branch=z9hG4bKZa7eS5j4r9a8F From: "0445674567" ;tag=vQ8mytQ2mXNme To: Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9c633c8 (303 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 100 Trying for INVITE (115960061) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 1.405 ms tport_release(0x9bb0e28): 0x9c63a58 by 0x9bcc770 with 0x9c633c8 (preliminary) nua(0x9cf7ed0): sent signal r_invite 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_INIT -> CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0631000001 at 192.168.0.2 entering state [calling][0] nua: nua_handle_magic: entering 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State INIT going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_ROUTING 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State ROUTING 2009-06-05 09:06:35 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/0631000001 at 192.168.0.2 SOFIA ROUTING 2009-06-05 09:06:35 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State ROUTING going to sleep 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_CONSUME_MEDIA 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State CONSUME_MEDIA 2009-06-05 09:06:35 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State CONSUME_MEDIA going to sleep nta: timer not set tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c633c8 from (udp/192.168.0.2:5070) has 714 bytes, veclen = 1 recv 714 bytes from udp/[192.168.0.2]:5060 at 23:06:35.925933: ------------------------------------------------------------------------ SIP/2.0 180 Ringing To: ;tag=f936c99a6bd026a9i0 From: "0445674567" ;tag=vQ8mytQ2mXNme Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Via: SIP/2.0/UDP 192.168.0.2:5070;received=192.168.0.2;rport=5070;branch=z9hG4bKZa7eS5j4r9a8F Record-Route: Record-Route: Record-Route: Server: Linksys/SPA3000-3.1.20(GW) Remote-Party-ID: 0631000001 ;screen=yes;party=called Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9c633c8 (714 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 180 Ringing for INVITE (115960061) nta: 180 Ringing is going to a transaction tport_release(0x9bb0e28): 0x9c63a58 by 0x9bcc770 with 0x9c633c8 (preliminary) nua(0x9cf7ed0): event r_invite 180 Ringing nua(0x9cf7ed0): call state changed: calling -> proceeding nua(0x9cf7ed0): event i_state 180 Ringing nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0631000001 at 192.168.0.2 entering state [proceeding][180] 2009-06-05 09:06:35 [NOTICE] sofia.c:3103 sofia_handle_sip_i_state() Ring-Ready sofia/internal/0631000001 at 192.168.0.2! 2009-06-05 09:06:35 [DEBUG] sofia.c:3111 sofia_handle_sip_i_state() sofia/internal/0445674567 at sip.evolutiontel.net receive message [RINGING] nua: nua_respond: entering nua(0x9ccc130): sent signal r_respond 2009-06-05 09:06:35 [NOTICE] mod_sofia.c:1422 sofia_receive_message() Ring-Ready sofia/internal/0445674567 at sip.evolutiontel.net! 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] nua: nua_handle_magic: entering nua(0x9ccc130): recv signal r_respond 180 Ringing nua: nua_stack_set_params: entering soa_set_params(static::0x9c1d4e8, ...) called nua: nua_invite_server_respond: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1125 bytes of 1125 to udp/192.168.0.2:5060 tport_vsend returned 1125 send 1125 bytes to udp/[192.168.0.2]:5060 at 23:06:35.927247: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Record-Route: Record-Route: From: "0445674567";tag=50f70f41-co453-INS001 To: ;tag=UeFvvZ6yQmZ1j Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE Contact: User-Agent: Evolutiontel SIP Service Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 P-Asserted-Identity: "" <0631000001> ------------------------------------------------------------------------ nta: sent 180 Ringing for INVITE (45301) nta: timer set to 60000 ms nua(0x9ccc130): call state changed: received -> early nua(0x9ccc130): event i_state 180 Ringing nua: nua_application_event: entering 2009-06-05 09:06:35 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [early][180] nua: nua_handle_magic: entering 2009-06-05 09:06:35 [DEBUG] switch_ivr_originate.c:1768 switch_ivr_originate() sofia/internal/0445674567 at sip.evolutiontel.net receive message [RINGING] 2009-06-05 09:06:35 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:35 [NOTICE] switch_ivr_originate.c:1768 switch_ivr_originate() Ring Ready sofia/internal/0445674567 at sip.evolutiontel.net! tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9d93fd8 from (udp/192.168.0.2:5070) has 1218 bytes, veclen = 1 recv 1218 bytes from udp/[192.168.0.2]:5060 at 23:06:41.842064: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=f936c99a6bd026a9i0 From: "0445674567" ;tag=vQ8mytQ2mXNme Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 INVITE Via: SIP/2.0/UDP 192.168.0.2:5070;received=192.168.0.2;rport=5070;branch=z9hG4bKZa7eS5j4r9a8F Record-Route: Record-Route: Record-Route: Contact: 0631000001 Server: Linksys/SPA3000-3.1.20(GW) Remote-Party-ID: 0631000001 ;screen=yes;party=called Content-Length: 306 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp P-Behind-NAT: Yes v=0 o=- 8503548 8503548 IN IP4 192.168.0.10 s=- c=IN IP4 60.xxx.xx.xxx t=0 0 m=audio 16582 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=direction:active a=oldmediaip:192.168.0.10 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9d93fd8 (1218 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for INVITE (115960061) nta: 200 OK is going to a transaction tport_release(0x9bb0e28): 0x9c63a58 by 0x9bcc770 with 0x9d93fd8 nta: timer shortened to 32000 ms soa_set_remote_sdp(static::0x9d15ce0, (nil), 0x9c62d18, 306) called soa_process_answer(static::0x9d15ce0) called soa_static_offer_answer_action(0x9d15ce0, soa_process_answer): called soa_sdp_mode_set(0x9cec6b0, 0x9db77b0, ""): called soa_static(0x9d15ce0, soa_process_answer): upgrade codecs with remote description soa_static(0x9d15ce0, soa_process_answer): storing local description soa_activate(static::0x9d15ce0, (nil)) called nua(0x9cf7ed0): INVITE: processed SDP answer in 200 OK nua(0x9cf7ed0): event r_invite 200 OK soa_activate(static::0x9d15ce0, (nil)) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 624 bytes of 624 to udp/192.168.0.2:5060 tport_vsend returned 624 send 624 bytes to udp/[192.168.0.2]:5060 at 23:06:41.842923: ------------------------------------------------------------------------ ACK sip:0631000001 at 60.xxx.xx.xxx:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5070;rport;branch=z9hG4bK0K07t037Nj1tB Route: Route: Route: Max-Forwards: 70 From: "0445674567" ;tag=vQ8mytQ2mXNme To: ;tag=f936c99a6bd026a9i0 Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960061 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ nta: sent ACK (115960061) to */192.168.0.2:5060 nua(0x9cf7ed0): call state changed: proceeding -> ready, received answer soa_get_remote_sdp(static::0x9d15ce0, [0xb77b7bfc], [0xb77b7bf8], [(nil)]) called soa_get_params(static::0x9d15ce0, ...) called nua(0x9cf7ed0): event i_state 200 OK nua(0x9cf7ed0): event i_active 200 Call active nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:41 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0631000001 at 192.168.0.2 entering state [ready][200] 2009-06-05 09:06:41 [DEBUG] sofia.c:3046 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 8503548 8503548 IN IP4 192.168.0.10 s=- c=IN IP4 60.xxx.xx.xxx t=0 0 m=audio 16582 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=direction:active a=oldmediaip:192.168.0.10 2009-06-05 09:06:41 [DEBUG] switch_channel.c:1875 switch_channel_perform_mark_answered() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:41 [DEBUG] switch_ivr_originate.c:1978 switch_ivr_originate() sofia/internal/0445674567 at sip.evolutiontel.net receive message [ANSWER] nua: nua_respond: entering nua(0x9ccc130): sent signal r_respond 2009-06-05 09:06:41 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:41 [NOTICE] switch_ivr_originate.c:1978 switch_ivr_originate() Channel [sofia/internal/0445674567 at sip.evolutiontel.net] has been answered 2009-06-05 09:06:41 [DEBUG] switch_channel.c:1911 switch_channel_perform_mark_answered() sofia/internal/0445674567 at sip.evolutiontel.net execute on answer: sched_hangup(+7200 ALLOTED_TIMEOUT) EXECUTE sofia/internal/0445674567 at sip.evolutiontel.net sched_hangup(+7200 ALLOTED_TIMEOUT) 2009-06-05 09:06:41 [DEBUG] switch_scheduler.c:214 switch_scheduler_add_task() Added task 7 switch_ivr_schedule_hangup (e57f4f54-7b45-48cc-98d3-8b594bdc19e8) to run at 1244164001 2009-06-05 09:06:41 [DEBUG] switch_ivr_originate.c:2024 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/0631000001 at 192.168.0.2] nua(0x9ccc130): recv signal r_respond 200 OK nua: nua_stack_set_params: entering soa_set_params(static::0x9c1d4e8, ...) called soa_set_user_sdp(static::0x9c1d4e8, (nil), 0x9dc2c08, -1) called soa_set_capability_sdp(static::0x9c1d4e8, (nil), 0x9dc2c08, -1) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x9c1d4e8) called soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called soa_static(0x9c1d4e8, soa_generate_answer): generating local description soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media soa_sdp_mode_set(0xb77b7fb4, 0x9ce6698, ""): called soa_init_sdp_connection_with_session: selected IN IP4 202.xxx.xx.xx (a local address) soa_static(0x9c1d4e8, soa_generate_answer): storing local description soa_activate(static::0x9c1d4e8, (nil)) called soa_get_local_sdp(static::0x9c1d4e8, [(nil)], [0xb77b803c], [0xb77b8038]) called tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 1414 bytes of 1414 to udp/192.168.0.2:5060 tport_vsend returned 1414 send 1414 bytes to udp/[192.168.0.2]:5060 at 23:06:41.847721: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-0 Record-Route: Record-Route: From: "0445674567";tag=50f70f41-co453-INS001 To: ;tag=UeFvvZ6yQmZ1j Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 INVITE Contact: User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 256 P-Asserted-Identity: "" <0631000001> v=0 o=- 2963411278722346302 7975655717233217142 IN IP4 202.xxx.xx.xx s=- c=IN IP4 60.xxx.xx.xxx t=0 0 m=audio 0 RTP/AVP 96 100 101 a=rtpmap:96 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (45301) nta: timer shortened to 500 ms nua(0x9ccc130): call state changed: early -> completed, sent answer soa_get_local_sdp(static::0x9c1d4e8, [0xb77b8124], [0xb77b8120], [(nil)]) called soa_get_params(static::0x9c1d4e8, ...) called nua(0x9ccc130): event i_state 200 OK 2009-06-05 09:06:41 [DEBUG] switch_ivr_bridge.c:791 switch_ivr_signal_bridge() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:41 [DEBUG] switch_ivr_bridge.c:792 switch_ivr_signal_bridge() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State EXECUTE going to sleep 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HIBERNATE 2009-06-05 09:06:41 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/0445674567 at sip.evolutiontel.net Standard HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HIBERNATE going to sleep 2009-06-05 09:06:41 [NOTICE] sofia.c:3509 sofia_handle_sip_i_state() Channel [sofia/internal/0631000001 at 192.168.0.2] has been answered nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:41 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [completed][200] nua: nua_handle_magic: entering 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HIBERNATE 2009-06-05 09:06:41 [DEBUG] mod_sofia.c:160 sofia_on_hibernate() sofia/internal/0631000001 at 192.168.0.2 SOFIA HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:212 switch_core_standard_on_hibernate() sofia/internal/0631000001 at 192.168.0.2 Standard HIBERNATE 2009-06-05 09:06:41 [DEBUG] switch_core_state_machine.c:505 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HIBERNATE going to sleep tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9cbc8b8 from (udp/192.168.0.2:5070) has 475 bytes, veclen = 1 recv 475 bytes from udp/[192.168.0.2]:5060 at 23:06:42.052636: ------------------------------------------------------------------------ ACK sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK92ec.85e18692.2 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f5-1 Max-Forwards: 69 To: ;tag=UeFvvZ6yQmZ1j From: "0445674567";tag=50f70f41-co453-INS001 Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45301 ACK User-Agent: ENSR2.5.4 Content-Length: 0 P-hint: rr-enforced ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9cbc8b8 (475 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received ACK sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 45301) nta: ACK (45301) is going to INVITE (45301) nua: process_ack_or_cancel: entering soa_clear_remote_sdp(static::0x9c1d4e8) called nua(0x9ccc130): event i_ack 200 OK nua(0x9ccc130): call state changed: completed -> ready nua(0x9ccc130): event i_state 200 OK nua(0x9ccc130): event i_active 200 Call active nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:42 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [ready][200] nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9cbc8b8 from (udp/192.168.0.2:5070) has 475 bytes, veclen = 1 recv 475 bytes from udp/[192.168.0.2]:5060 at 23:06:42.088227: ------------------------------------------------------------------------ BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK62ec.b694c4f3.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f6-0 Max-Forwards: 69 To: ;tag=UeFvvZ6yQmZ1j From: "0445674567";tag=50f70f41-co453-INS001 Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45302 BYE User-Agent: ENSR2.5.4 Content-Length: 0 P-hint: rr-enforced ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9cbc8b8 (475 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received BYE sip:mod_sofia at 192.168.0.2:5070 SIP/2.0 (CSeq 45302) nta: canonizing sip:mod_sofia at 192.168.0.2:5070 with contact nta: BYE (45302) going to existing leg nua: nua_stack_process_request: entering tport_tsend(0x9bb0e28) tpn = UDP/192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name UDP/192.168.0.2:5060 tport_vsend(0x9bb0e28): 588 bytes of 588 to udp/192.168.0.2:5060 tport_vsend returned 588 send 588 bytes to udp/[192.168.0.2]:5060 at 23:06:42.088632: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK62ec.b694c4f3.0 Via: SIP/2.0/UDP 202.xx.xxx.xx:5060;branch=z9hG4bK1ca53b7214a28536c-b0f6-0 From: "0445674567";tag=50f70f41-co453-INS001 To: ;tag=UeFvvZ6yQmZ1j Call-ID: 2619-475-54200923635-IMG01-0-202.83.183.46 CSeq: 45302 BYE User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for BYE (45302) nua(0x9ccc130): event i_bye 200 Session Terminated nua(0x9ccc130): removing session usage nua(0x9ccc130): call state changed: ready -> terminated nua(0x9ccc130): event i_state 200 Session Terminated nua(0x9ccc130): event i_terminated 200 Session Terminated soa_destroy(static::0x9c1d4e8) called nta_leg_destroy(0x9c41a68) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2009-06-05 09:06:42 [DEBUG] sofia.c:3039 sofia_handle_sip_i_state() Channel sofia/internal/0445674567 at sip.evolutiontel.net entering state [terminated][200] 2009-06-05 09:06:42 [NOTICE] sofia.c:3599 sofia_handle_sip_i_state() Hangup sofia/internal/0445674567 at sip.evolutiontel.net [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-05 09:06:42 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [KILL] 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua(0x9ccc130): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nua: nua_application_event: entering nua(0x9ccc130): event i_terminated dropped 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_HANGUP 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HANGUP 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/0445674567 at sip.evolutiontel.net Overriding SIP cause 480 with 200 from the other leg 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/0445674567 at sip.evolutiontel.net hanging up, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [NOTICE] switch_ivr_bridge.c:712 signal_bridge_on_hangup() Hangup sofia/internal/0631000001 at 192.168.0.2 [CS_HIBERNATE] [NORMAL_CLEARING] 2009-06-05 09:06:42 [DEBUG] switch_channel.c:1667 switch_channel_perform_hangup() Send signal sofia/internal/0631000001 at 192.168.0.2 [KILL] 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/0445674567 at sip.evolutiontel.net Standard HANGUP, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State HANGUP going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_HANGUP -> CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0445674567 at sip.evolutiontel.net [BREAK] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) Running State Change CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State REPORTING nua(0x9ccc130): recv signal r_destroy nta_leg_destroy((nil)) 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_HANGUP 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HANGUP 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:307 sofia_on_hangup() sofia/internal/0631000001 at 192.168.0.2 Overriding SIP cause 480 with 200 from the other leg 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:339 sofia_on_hangup() Channel sofia/internal/0631000001 at 192.168.0.2 hanging up, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:394 sofia_on_hangup() Sending BYE to sofia/internal/0631000001 at 192.168.0.2 nua: nua_bye: entering nua(0x9cf7ed0): sent signal r_bye 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/0631000001 at 192.168.0.2 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State HANGUP going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_HANGUP -> CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000001 at 192.168.0.2 [BREAK] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) Running State Change CS_REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0631000001 at 192.168.0.2) State REPORTING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/0631000001 at 192.168.0.2 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0631000001 at 192.168.0.2) State REPORTING going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/0631000001 at 192.168.0.2) State Change CS_REPORTING -> CS_DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 10 (sofia/internal/0631000001 at 192.168.0.2) Locked, Waiting on external entities 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 10 (sofia/internal/0631000001 at 192.168.0.2) Ended 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/0631000001 at 192.168.0.2 [CS_DESTROY] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0631000001 at 192.168.0.2) State DESTROY 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/0631000001 at 192.168.0.2 SOFIA DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/0631000001 at 192.168.0.2 Standard DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0631000001 at 192.168.0.2) State DESTROY going to sleep nua(0x9cf7ed0): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x9d15ce0, ...) called soa_terminate(static::0x9d15ce0) called soa_init_offer_answer(static::0x9d15ce0) called nta: selecting scheme sip tport_tsend(0x9bb0e28) tpn = */192.168.0.2:5060 tport_resolve addrinfo = 192.168.0.2:5060 tport_by_addrinfo(0x9bb0e28): not found by name */192.168.0.2:5060 tport_vsend(0x9bb0e28): 874 bytes of 874 to udp/192.168.0.2:5060 tport_vsend returned 874 send 874 bytes to udp/[192.168.0.2]:5060 at 23:06:42.094504: ------------------------------------------------------------------------ BYE sip:0631000001 at 60.xxx.xx.xxx:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5070;rport;branch=z9hG4bK1vS0vUmBKUQDQ Route: Route: Route: Max-Forwards: 70 From: "0445674567" ;tag=vQ8mytQ2mXNme To: ;tag=f936c99a6bd026a9i0 Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960062 BYE Contact: User-Agent: Evolutiontel SIP Service Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ nta: sent BYE (115960062) to */192.168.0.2:5060 tport_pend(0x9bb0e28): pending 0x9d15e08 for udp/192.168.0.2:5070 (already 0) 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/0445674567 at sip.evolutiontel.net Standard REPORTING, cause: NORMAL_CLEARING 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State REPORTING going to sleep 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/0445674567 at sip.evolutiontel.net) State Change CS_REPORTING -> CS_DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 9 (sofia/internal/0445674567 at sip.evolutiontel.net) Locked, Waiting on external entities 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 9 (sofia/internal/0445674567 at sip.evolutiontel.net) Ended 2009-06-05 09:06:42 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/0445674567 at sip.evolutiontel.net [CS_DESTROY] 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State DESTROY 2009-06-05 09:06:42 [DEBUG] mod_sofia.c:256 sofia_on_destroy() sofia/internal/0445674567 at sip.evolutiontel.net SOFIA DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/0445674567 at sip.evolutiontel.net Standard DESTROY 2009-06-05 09:06:42 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0445674567 at sip.evolutiontel.net) State DESTROY going to sleep tport_wakeup_pri(0x9bb0e28): events IN tport_recv_event(0x9bb0e28) tport_recv_iovec(0x9bb0e28) msg 0x9c41b38 from (udp/192.168.0.2:5070) has 450 bytes, veclen = 1 recv 450 bytes from udp/[192.168.0.2]:5060 at 23:06:42.210621: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=f936c99a6bd026a9i0 From: "0445674567" ;tag=vQ8mytQ2mXNme Call-ID: 315cd1e2-cbff-122c-ee95-00112fcc990a CSeq: 115960062 BYE Via: SIP/2.0/UDP 192.168.0.2:5070;received=192.168.0.2;rport=5070;branch=z9hG4bK1vS0vUmBKUQDQ Server: Linksys/SPA3000-3.1.20(GW) P-RTP-Stat: PS=12,OS=240,PR=0,OR=0,PL=0,JI=0,LA=0,DU=0,EN=G729a,DE=G711u Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x9bb0e28): msg 0x9c41b38 (450 bytes) from udp/192.168.0.2:5070/sip next=(nil) nta: received 200 OK for BYE (115960062) nta: 200 OK is going to a transaction nta_outgoing: RTT is 116.446 ms tport_release(0x9bb0e28): 0x9d15e08 by 0x9bcdcb0 with 0x9c41b38 nua(0x9cf7ed0): event r_bye 200 OK nua(0x9cf7ed0): call state changed: terminating -> terminated nua(0x9cf7ed0): event i_state 200 to BYE nua(0x9cf7ed0): event i_terminated 200 to BYE nua(0x9cf7ed0): removing session usage soa_destroy(static::0x9d15ce0) called nta_leg_destroy(0x9db94b0) nua: terminated session 0x9cf7ed0 nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x9cf7ed0): sent signal r_destroy nua: nua_application_event: entering nua(0x9cf7ed0): event i_terminated dropped nua(0x9cf7ed0): recv signal r_destroy nta_leg_destroy((nil)) nta: timer set next to 4704 ms 2009-06-05 09:06:42 [DEBUG] switch_scheduler.c:138 task_thread_loop() Deleting task 7 switch_ivr_schedule_hangup (e57f4f54-7b45-48cc-98d3-8b594bdc19e8) nta: timer I fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0xb77b81cc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 157 ms nta: timer K fired, terminate BYE (115960062) outgoing_reclaim_all((nil), (nil), 0xb77b81c8) nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/2 term, 1/3 free nta: timer set next to 26628 ms /exit [root at sip01 bin]# From jim at evolutiontel.net Thu Jun 4 16:18:36 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 09:18:36 +1000 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: <079f01c9e535$c1fed2e0$45fc78a0$@com> References: <079f01c9e535$c1fed2e0$45fc78a0$@com> Message-ID: Not that this helps your question directly. Using a putty terminal from windows allows the data to be copied correctly from console sessions. On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: > I want to copy the results of a siptrace captured on the fs console to a > file. The console is running on a Gnome terminal. I highlight the text I > want to copy in the fs console, open a vi session in insert mode, and paste > the text. However, the text is not pasted as I copied it ? it is missing > characters/lines. > > > > I know I am doing something wrong. Is there another way to save siptraces to > a file? Redirection doesn?t work. > > > > sofia profile internal siptrace on is the command I use. > > > > Thanks Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From brian at freeswitch.org Thu Jun 4 16:28:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 18:28:57 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> Message-ID: <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> Port 0 indicates a rejection. Something is sending a 200 OK with 0 and g729a on codec 96. Have you modified the freeswitch code ? /b On Jun 4, 2009, at 6:15 PM, Jim Burke wrote: > Hi Anthony, > > Traces as requested. Let me know if you want a jira opened or any > further data. > > Regards, > Jim Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0e46b828/attachment-0002.html From msc at freeswitch.org Thu Jun 4 16:29:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 16:29:18 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> Message-ID: <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood wrote: > Okay, leave this with me - I'll bring this together and do what's required. > I'll see if I can squeeze it in ahead of schedule, but don't count on it. > Also expect pleas for help at some stage, and I'll need a UI developer to > create an awesome interface. > > You'll all be hearing from me shortly! > You're hired!!! :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/5b561c5c/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 16:36:26 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 16:36:26 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> Message-ID: <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> Awesome :-D I need some thinking time to come up with a strategy, but I'll set up a project site soon so we can all collaborate on this. M On Thu, Jun 4, 2009 at 4:29 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> Okay, leave this with me - I'll bring this together and do what's >> required. I'll see if I can squeeze it in ahead of schedule, but don't count >> on it. Also expect pleas for help at some stage, and I'll need a UI >> developer to create an awesome interface. >> >> You'll all be hearing from me shortly! >> > > You're hired!!! :) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/f2d9ecce/attachment-0002.html From msc at freeswitch.org Thu Jun 4 16:47:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 16:47:20 -0700 Subject: [Freeswitch-users] Getting Ready To Welcome The Newest Little FreeSWITCH Community Member! Message-ID: <87f2f3b90906041647w727e04c0je357c2e2bad32de7@mail.gmail.com> Hey everyone, Raymond Chandler (aka Intralanman) and his wife are expecting a baby boy! Check out the story and how you can help welcome our newest FreeSWITCH community member: http://www.freeswitch.org/node/189 -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/9d75f6df/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 16:49:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 09:49:39 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> Message-ID: Using FreeSWITCH Version 1.0.trunk (13523) and have not modified the code. Yes, exactly this is what causes the originator to release the call. As you will see in the traces, the 200OK is good on the way into FS, but looks to be trampled on the way out :( Regards, On Fri, Jun 5, 2009 at 9:28 AM, Brian West wrote: > Port 0 indicates a rejection. ?Something is sending a 200 OK with 0 and > g729a on codec 96. ?Have you modified the freeswitch code ? > /b > On Jun 4, 2009, at 6:15 PM, Jim Burke wrote: > > Hi Anthony, > > Traces as requested. ?Let me know if you want a jira opened or any further > data. > > Regards, > Jim > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From brian at freeswitch.org Thu Jun 4 16:51:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 18:51:13 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> Message-ID: <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> But that SDP is not generated by FreeSWITCH... are you using proxy media or something? maybe bypass? /b On Jun 4, 2009, at 6:49 PM, Jim Burke wrote: > Yes, exactly this is what causes the originator to release the call. > As you will see in the traces, the 200OK is good on the way into FS, > but looks to be trampled on the way out :( > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/371a927e/attachment-0002.html From d at unwire.it Thu Jun 4 16:58:23 2009 From: d at unwire.it (Darin Weeks) Date: Thu, 4 Jun 2009 16:58:23 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> Message-ID: <989132e70906041658xba4f660u91ae3793f581f327@mail.gmail.com> I'm game to help out. I have "product" experience with UI and stuff... (but please don't think that I'm a business wonk). As a new freeswitch user it might be slightly harder to work with me rather than someone who knows the nuts and bolts of the system, but on the other hand, my n00b status might yield a more intuitive UI because it'll have to make sense to me in order to design it. Please keep me in the loop or add me to any emails on the topic and I'll see how I can join in. On Thu, Jun 4, 2009 at 4:36 PM, Matthew Lockwood wrote: > Awesome :-D > > I need some thinking time to come up with a strategy, but I'll set up a > project site soon so we can all collaborate on this. > > M > > On Thu, Jun 4, 2009 at 4:29 PM, Michael Collins wrote: > >> >> >> On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood < >> matthew.lockwood at gmail.com> wrote: >> >>> Okay, leave this with me - I'll bring this together and do what's >>> required. I'll see if I can squeeze it in ahead of schedule, but don't count >>> on it. Also expect pleas for help at some stage, and I'll need a UI >>> developer to create an awesome interface. >>> >>> You'll all be hearing from me shortly! >>> >> >> You're hired!!! :) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/50993fcd/attachment-0002.html From matthew.lockwood at gmail.com Thu Jun 4 17:03:26 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 17:03:26 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <989132e70906041658xba4f660u91ae3793f581f327@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <87f2f3b90906041629v51863932vb12af9dd14b9086b@mail.gmail.com> <415541b10906041636n5927d559yf3207101662e77ac@mail.gmail.com> <989132e70906041658xba4f660u91ae3793f581f327@mail.gmail.com> Message-ID: <415541b10906041703s6f1e6a1dwaf0930023b401679@mail.gmail.com> Okay great. What I'll be doing is creating the back-end. When the project kicks off, it would be far better for someone else to design a pretty and useable UI on top of that. You'd not really need to know much about the inner workings of FS. I'll probably just end up knocking up a fugly unstyled HTML interface for testing purposes, so you could just take it from there. I'll be the business wonk! I'll be setting up a project site on my company's intranet soon. On Thu, Jun 4, 2009 at 4:58 PM, Darin Weeks wrote: > I'm game to help out. I have "product" experience with UI and stuff... > (but please don't think that I'm a business wonk). As a new freeswitch user > it might be slightly harder to work with me rather than someone who knows > the nuts and bolts of the system, but on the other hand, my n00b status > might yield a more intuitive UI because it'll have to make sense to me in > order to design it. > > Please keep me in the loop or add me to any emails on the topic and I'll > see how I can join in. > > > On Thu, Jun 4, 2009 at 4:36 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> Awesome :-D >> >> I need some thinking time to come up with a strategy, but I'll set up a >> project site soon so we can all collaborate on this. >> >> M >> >> On Thu, Jun 4, 2009 at 4:29 PM, Michael Collins wrote: >> >>> >>> >>> On Thu, Jun 4, 2009 at 4:13 PM, Matthew Lockwood < >>> matthew.lockwood at gmail.com> wrote: >>> >>>> Okay, leave this with me - I'll bring this together and do what's >>>> required. I'll see if I can squeeze it in ahead of schedule, but don't count >>>> on it. Also expect pleas for help at some stage, and I'll need a UI >>>> developer to create an awesome interface. >>>> >>>> You'll all be hearing from me shortly! >>>> >>> >>> You're hired!!! :) >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/e43c4748/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 17:07:44 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 10:07:44 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> Message-ID: bypass_media is set, and proxy_media is not set. I agree, FS should not be touching the SDP for calls in bypass_media mode. Interesting, when you look in the file when FS reports the Remote SDP it still looks ok, then a little further down you can see it send the 200OK out to the originator and that SDP in-correctly reports the media port. The other interesting point is a=rtpmap:96 G729a/8000. 96 is not the correct rtpmap for G729 and it is not mentioned on the incoming 200Ok to FS Regards, On Fri, Jun 5, 2009 at 9:51 AM, Brian West wrote: > But that SDP is not generated by FreeSWITCH... are you using proxy media or > something? ?maybe bypass? > /b > On Jun 4, 2009, at 6:49 PM, Jim Burke wrote: > > Yes, exactly this is what causes the originator to release the call. > As you will see in the traces, the 200OK is good on the way into FS, > but looks to be trampled on the way out :( > > Regards, > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From jim at evolutiontel.net Thu Jun 4 17:23:36 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 10:23:36 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> Message-ID: Hey Brian >From your comments above this appears to be the code that does the damage. I guess now the question is why?? soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called soa_static(0x9c1d4e8, soa_generate_answer): generating local description soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media Regards, On Fri, Jun 5, 2009 at 10:07 AM, Jim Burke wrote: > bypass_media is set, and proxy_media is not set. > > I agree, FS should not be touching the SDP for calls in bypass_media > mode. ?Interesting, when you look in the file when FS reports the > Remote SDP it still looks ok, then a little further down you can see > it send the 200OK out to the originator and that SDP in-correctly > reports the media port. > > The other interesting point is a=rtpmap:96 G729a/8000. ?96 is not the > correct rtpmap for G729 and it is not mentioned on the incoming 200Ok > to FS > > Regards, > > > On Fri, Jun 5, 2009 at 9:51 AM, Brian West wrote: >> But that SDP is not generated by FreeSWITCH... are you using proxy media or >> something? ?maybe bypass? >> /b >> On Jun 4, 2009, at 6:49 PM, Jim Burke wrote: >> >> Yes, exactly this is what causes the originator to release the call. >> As you will see in the traces, the 200OK is good on the way into FS, >> but looks to be trampled on the way out :( >> >> Regards, >> >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! ?http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From brian at freeswitch.org Thu Jun 4 17:34:17 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 19:34:17 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> Message-ID: <53828315-70FE-4A7E-A0B0-3608E16FCF56@freeswitch.org> Try SVN trunk I cna tell you're using older code! ;) /b On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: > Hey Brian > >> From your comments above this appears to be the code that does the > damage. I guess now the question is why?? > > soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called > soa_static(0x9c1d4e8, soa_generate_answer): generating local > description > soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote > description > soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/e7ba7835/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 18:00:28 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 11:00:28 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. In-Reply-To: <53828315-70FE-4A7E-A0B0-3608E16FCF56@freeswitch.org> References: <191c3a030906040559v423f5777t89ba19982705d851@mail.gmail.com> <56A5C697-36A9-4CCD-BA0C-F79ECB90C612@freeswitch.org> <023F5603-4DB1-4550-9D74-87A22EC364D9@freeswitch.org> <53828315-70FE-4A7E-A0B0-3608E16FCF56@freeswitch.org> Message-ID: Hmmm...no luck with the SVN trunk FreeSWITCH Version 1.0.trunk (13624) On Fri, Jun 5, 2009 at 10:34 AM, Brian West wrote: > Try SVN trunk I cna tell you're using older code! ?;) > /b > On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: > > Hey Brian > > From your comments above this appears to be the code that does the > > damage. ?I guess now the question is why?? > > soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called > soa_static(0x9c1d4e8, soa_generate_answer): generating local description > soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description > soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media > > Regards, > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From dujinfang at gmail.com Thu Jun 4 18:38:46 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 09:38:46 +0800 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: <079f01c9e535$c1fed2e0$45fc78a0$@com> References: <079f01c9e535$c1fed2e0$45fc78a0$@com> Message-ID: <1E2BD79E-D1B8-41D1-8560-791F25A68E5A@gmail.com> try this: copy the text open a new terminal cat > siptrace.txt paste the text press [Enter] press [Ctrl] + d On Jun 5, 2009, at 12:59 AM, Lars Zeb wrote: > I want to copy the results of a siptrace captured on the fs console > to a file. The console is running on a Gnome terminal. I highlight > the text I want to copy in the fs console, open a vi session in > insert mode, and paste the text. However, the text is not pasted as > I copied it ? it is missing characters/lines. > > I know I am doing something wrong. Is there another way to save > siptraces to a file? Redirection doesn?t work. > > sofia profile internal siptrace on is the command I use. > > Thanks Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/9117d3c8/attachment-0002.html From larclap at yahoo.com Thu Jun 4 18:38:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 4 Jun 2009 18:38:45 -0700 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: References: <079f01c9e535$c1fed2e0$45fc78a0$@com> Message-ID: <08bc01c9e57e$5ef60030$1ce20090$@com> Does this mean that I must start fs from the putty terminal, or can I attach to an already running instance via putty? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim Burke Sent: Thursday, June 04, 2009 4:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing lines copying data from console to vi Not that this helps your question directly. Using a putty terminal from windows allows the data to be copied correctly from console sessions. On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: > I want to copy the results of a siptrace captured on the fs console to a > file. The console is running on a Gnome terminal. I highlight the text I > want to copy in the fs console, open a vi session in insert mode, and paste > the text. However, the text is not pasted as I copied it - it is missing > characters/lines. > > > > I know I am doing something wrong. Is there another way to save siptraces to > a file? Redirection doesn't work. > > > > sofia profile internal siptrace on is the command I use. > > > > Thanks Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From intralanman at freeswitch.org Thu Jun 4 18:40:12 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 04 Jun 2009 21:40:12 -0400 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906041539r47bdf2c6m1df4671ad4f67342@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041439g46d34f90j66b1195df1166717@mail.gmail.com> <87f2f3b90906041539r47bdf2c6m1df4671ad4f67342@mail.gmail.com> Message-ID: <4A28777C.1000609@freeswitch.org> we created a #freeswitch-gui channel for talks such as these... I'm the only one idling in it most of the time, though :-( -Ray Michael Collins wrote: > > On Thu, Jun 4, 2009 at 2:39 PM, Matthew Lockwood > > wrote: > > I'm about to start creating one. I think FS needs a UI comparable > to FreePBX or similar. It's the next project on my list. It would > certainly be a boost to the FS project and user acceptance. > > > Be sure to check with Bougyman on IRC. He is planning on releasing an > open-source, MIT-licensed FS-GUI w/ underlying framework. Last I heard > he said week of June 15. Note: he said that it will require Ruby plus > Rack and PostgreSQL. > > -MC > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/8c050b6a/attachment-0002.html From edpimentl at gmail.com Thu Jun 4 18:47:54 2009 From: edpimentl at gmail.com (EdPimentl) Date: Thu, 4 Jun 2009 21:47:54 -0400 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A2847BB.8020306@freeswitch.org> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> Message-ID: <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> I would like to see it, and you would get many more volunteers if it would jquery/mootools UI with DJANGO backend. Or a MOZILLA XUL Or a DOJO APP Or a FLEX/AIR That would be different and unique.... -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/aa623596/attachment-0002.html From jim at evolutiontel.net Thu Jun 4 18:51:32 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 11:51:32 +1000 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: <08bc01c9e57e$5ef60030$1ce20090$@com> References: <079f01c9e535$c1fed2e0$45fc78a0$@com> <08bc01c9e57e$5ef60030$1ce20090$@com> Message-ID: After logging into my linux box using putty. I then change directory the the ~/freeswitch/bin directory and run ./fs_cli You can, but you don't need to start FS from the putty terminal. We run ours as a background process. On Fri, Jun 5, 2009 at 11:38 AM, Lars Zeb wrote: > Does this mean that I must start fs from the putty terminal, or can I attach > to an already running instance via putty? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim > Burke > Sent: Thursday, June 04, 2009 4:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Missing lines copying data from console to > vi > > Not that this helps your question directly. ?Using a putty terminal > from windows allows the data to be copied correctly from console > sessions. > > > > On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: >> I want to copy the results of a siptrace captured on the fs console to a >> file. The console is running on a Gnome terminal. I highlight the text I >> want to copy in the fs console, open a vi session in insert mode, and > paste >> the text. However, the text is not pasted as I copied it - it is missing >> characters/lines. >> >> >> >> I know I am doing something wrong. Is there another way to save siptraces > to >> a file? Redirection doesn't work. >> >> >> >> sofia profile internal siptrace on is the command I use. >> >> >> >> Thanks Lars >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From matthew.lockwood at gmail.com Thu Jun 4 18:52:51 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Thu, 4 Jun 2009 18:52:51 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <415541b10906041523y469ff381t87a7760d9d5d65c1@mail.gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> Message-ID: <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. On Thu, Jun 4, 2009 at 6:47 PM, EdPimentl wrote: > I would like to see it, and you would get many more volunteers if it would > jquery/mootools UI with DJANGO backend. > > Or a MOZILLA XUL > > Or a DOJO APP > > Or a FLEX/AIR > > That would be different and unique.... > -E > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/dfe9dc94/attachment-0002.html From dujinfang at gmail.com Thu Jun 4 18:59:10 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 09:59:10 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <200906041628066407689@gmail.com> References: <200906041628066407689@gmail.com> Message-ID: <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then. On Jun 4, 2009, at 4:28 PM, god.nirvana wrote: > hi all > i am new to freeswitch. > there are some busy tone detect issues,i hope someone could help me. > i installed freeswitch from trunk,openzap,zaptel.... > but i found some busy tone isuues > > my tones.conf: > [us] > generate-dial => v=-7;%(1000,0,350,440) > detect-dial => 350,440 > generate-ring => v=-7;%(2000,4000,440,480) > detect-ring => 440,480 > generate-busy => v=-7;%(500,500,450,340) > detect-busy =>450,340 > generate-attn => v=0;%(100,100,1400,2060,2450,2600) > detect-attn => 1400,2060,2450,2600 > generate-callwaiting-sas => v=0;%(300,0,440) > detect-callwaiting-sas => 440 > generate-callwaiting-cas => v=0;%(80,0,2750,2130) > detect-callwaiting-cas => 2750,2130 > detect-fail1 => 913.8 > detect-fail2 => 1370.6 > detect-fail3 => 776.7 > > > openzap.conf.xml : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > when i call the pstn phone from a ip phone,if the pstn call hangup > first,the ip phone will hear the busy tone,but the system does not > handle the busytone ,the channel does not erase. so i have to add > in the dialplan.and it works.the channel > erased. > > but in the conference case,pstn phone call in,hangup. all > participants hear the tone,"do ~,do~~".freeswitch doest handle it. > so i change the conference dialplan. > > > > > > > > > > restart freeswitch,try again,freeswitch not handle the hangup tone > still,all participants hear the tone. > how to solve it?could some one help me ??? > thx! > BR > M > .Q > 2009-06-04 > god.nirvana > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/dd27b14e/attachment-0002.html From brad.tuan at gmail.com Thu Jun 4 19:43:40 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Fri, 5 Jun 2009 10:43:40 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: Yes I have tried it,but useless, when 1001 and 1002 are talking to each other , then 1001 want to transfer 1002 to 1003, so 1001 press *1 1003, but nothing happen....... 2009/6/4 dujinfang > yes. Did you ever tried that? > > On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: > > I mean when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003. > 2009/6/4 seven > >> the default config allows 1002 press *1 and 1003 to do blind transfer, >> also you may interest the att_xfer, see dp_tools on wiki. >> >> >> On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >> >> If i don't want to use softphone function to transfer the call ,how to do >> it?? >> >> 2009/6/4 Brian West >> >>> Depends.. Press the transfer key on your phone is how I would do it.. >>> what kind of phone do you have? >>> /b >>> >>> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >>> >>> When User(1001) calling with User(1002) , >>> >>> how to transfer User(1002) to User(1003)?? >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/c6147ea8/attachment-0002.html From brian at freeswitch.org Thu Jun 4 19:50:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 4 Jun 2009 21:50:13 -0500 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: Please show us your example of using att_xfr /b On Jun 4, 2009, at 9:43 PM, Brad Tuan wrote: > Yes I have tried it,but useless, > > when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003, > > so 1001 press *1 1003, > > but nothing happen....... Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/c1473300/attachment-0002.html From dujinfang at gmail.com Thu Jun 4 20:09:12 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 11:09:12 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: guess if you press *1 on 1002 you can transfer 1001 to 1003 if you want to press *1 on 1001, find Local_extension in dianplan/ default.xml where data = "0 b ... means it's only works on the b-leg try change to a (for a-leg) or ab for both leg. check bind_meta_app for detail on wiki, I bet you never tried the att_xfer feature. On Jun 5, 2009, at 10:43 AM, Brad Tuan wrote: > Yes I have tried it,but useless, > > when 1001 and 1002 are talking to each other , > > then 1001 want to transfer 1002 to 1003, > > so 1001 press *1 1003, > > but nothing happen....... > 2009/6/4 dujinfang > yes. Did you ever tried that? > > On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: >> I mean when 1001 and 1002 are talking to each other , >> >> then 1001 want to transfer 1002 to 1003. >> 2009/6/4 seven >> the default config allows 1002 press *1 and 1003 to do blind >> transfer, also you may interest the att_xfer, see dp_tools on wiki. >> >> >> On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >>> If i don't want to use softphone function to transfer the >>> call ,how to do it?? >>> >>> 2009/6/4 Brian West >>> Depends.. Press the transfer key on your phone is how I would do >>> it.. what kind of phone do you have? >>> >>> /b >>> >>> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >>> >>>> When User(1001) calling with User(1002) , >>>> >>>> how to transfer User(1002) to User(1003)?? >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/d62d6649/attachment-0002.html From brad.tuan at gmail.com Thu Jun 4 21:37:27 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Fri, 5 Jun 2009 12:37:27 +0800 Subject: [Freeswitch-users] A problem of call transfer In-Reply-To: References: <8614A03D-6DBF-45D6-8D48-FB370F01A4DB@gmail.com> Message-ID: Yes I got it is the call leg setting problem,thanks 2009/6/5 seven > guess if you press *1 on 1002 you can transfer 1001 to 1003 > if you want to press *1 on 1001, find Local_extension in > dianplan/default.xml > > where data = "0 b ... means it's only works on the b-leg > > try change to a (for a-leg) or ab for both leg. > > > > check bind_meta_app for detail on wiki, I bet you never tried the att_xfer > feature. > > On Jun 5, 2009, at 10:43 AM, Brad Tuan wrote: > > Yes I have tried it,but useless, > > when 1001 and 1002 are talking to each other , > then 1001 want to transfer 1002 to 1003, > > so 1001 press *1 1003, > > but nothing happen....... > 2009/6/4 dujinfang > >> yes. Did you ever tried that? >> >> On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote: >> >> I mean when 1001 and 1002 are talking to each other , >> >> then 1001 want to transfer 1002 to 1003. >> 2009/6/4 seven >> >>> the default config allows 1002 press *1 and 1003 to do blind transfer, >>> also you may interest the att_xfer, see dp_tools on wiki. >>> >>> >>> On Jun 4, 2009, at 4:14 PM, Brad Tuan wrote: >>> >>> If i don't want to use softphone function to transfer the call ,how to do >>> it?? >>> >>> 2009/6/4 Brian West >>> >>>> Depends.. Press the transfer key on your phone is how I would do it.. >>>> what kind of phone do you have? >>>> /b >>>> >>>> On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote: >>>> >>>> When User(1001) calling with User(1002) , >>>> >>>> how to transfer User(1002) to User(1003)?? >>>> >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/573039a1/attachment-0002.html From msc at freeswitch.org Thu Jun 4 22:14:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 22:14:50 -0700 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> Message-ID: <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > I'm using openzap analog with tone_detect, it works(conference not tested). > however, according to the asterisk book, Kewlstart can detect the busy tone > and disconnect the circuit. does anyone knows how to configure kewlstart > with freeswitch/openzap? guess we don't need tone_detect then. > Dujinfang, Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.) Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/417cbbd8/attachment-0002.html From stevecrozz at gmail.com Thu Jun 4 22:16:40 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 4 Jun 2009 22:16:40 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> Message-ID: <11990ade0906042216p13f8b05au778cb12b379ef98a@mail.gmail.com> In case anyone was wondering, I could lend a hand if django/python, ruby on rails, or php on were involved. --Stephen On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood wrote: > That would involve me learning a totally new framework. It'll not the > hardest code I'll ever write by far, so I'm okay coding it up on my own. > However, I definitely need a lot of help from fabulous designers to actually > make the interface pretty and useable. Plus, I'm only one person and will > need a lot of feedback to create something that rocks - everybody has a > different use case and I can't foresee how everybody will use it, so that > kind of feedback will go into re-engineering it. > > On Thu, Jun 4, 2009 at 6:47 PM, EdPimentl wrote: > >> I would like to see it, and you would get many more volunteers if it would >> jquery/mootools UI with DJANGO backend. >> >> Or a MOZILLA XUL >> >> Or a DOJO APP >> >> Or a FLEX/AIR >> >> That would be different and unique.... >> -E >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0a99f98e/attachment-0002.html From msc at freeswitch.org Thu Jun 4 22:36:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 4 Jun 2009 22:36:05 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> Message-ID: <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood wrote: > That would involve me learning a totally new framework. It'll not the > hardest code I'll ever write by far, so I'm okay coding it up on my own. > However, I definitely need a lot of help from fabulous designers to actually > make the interface pretty and useable. Plus, I'm only one person and will > need a lot of feedback to create something that rocks - everybody has a > different use case and I can't foresee how everybody will use it, so that > kind of feedback will go into re-engineering it. > If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited opinions about all sorts of things. :) All that being said, I say go for it. Find what works for you and see what happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure that we could even start a mailing list for GUI development. Enjoy! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090604/0033a8f9/attachment-0002.html From jim at evolutiontel.net Fri Jun 5 01:06:41 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 5 Jun 2009 18:06:41 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. Message-ID: <3wmhHdntwTK5.tvBsy3CM@smtp.gmail.com> should open a jira on this? Did you guys need any more info? - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. From: Brian West Date: 05/06/2009 00:35 Try SVN trunk I cna tell you're using older code! ;) /b On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: > Hey Brian > >> From your comments above this appears to be the code that does the > damage. I guess now the question is why?? > > soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called > soa_static(0x9c1d4e8, soa_generate_answer): generating local > description > soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote > description > soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Fri Jun 5 01:26:11 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 16:26:11 +0800 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com> <415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> Message-ID: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood > wrote: > That would involve me learning a totally new framework. It'll not > the hardest code I'll ever write by far, so I'm okay coding it up on > my own. However, I definitely need a lot of help from fabulous > designers to actually make the interface pretty and useable. Plus, > I'm only one person and will need a lot of feedback to create > something that rocks - everybody has a different use case and I > can't foresee how everybody will use it, so that kind of feedback > will go into re-engineering it. > > If you guys are serious about this then I would like to make a few > suggestions that might be obvious but for the sake of the project > we'll make them explicitly obvious. > > First, before deciding what framework to use, it would be good to > hold some discussions about what the GUI actually needs to do: > What are the design goals? > Will it be just for setting up extensions and the dialplan? Or will > it go much farther than that? > Will you be using mod_xml_curl for everything? If so, what > database(s) will you support? > Are you going to have extra goodies like an IVR builder? > A 'visual voicemail' page? > A user portal? > Management interface to 'spy' on users? > A CDR/call accounting system? > FIFO and/or ACD queue management? > MOH and sound files management? and conference management > > > It's okay to start small and build your way out, but you need to > know before you start building what the grand scheme will be. The > larger the goals of the project, the narrower your choices for a > framework that can do it all. The simple fact of the matter is that > if you want to use a MVC web framework then you have a somewhat > limited number of choices. You need a MVC WF that fits your needs, > which means it needs to be at least somewhat flexible. If you want a > pretty GUI then you need to decide if you want a rich Internet > application (RIA) front end like AIR, or do you want something along > the lines of XHTML/CSS/JS and use a platform like Dojo which gives > you cross-browser widgets and tools. All of this on top of the fact > that if you want volunteers to assist you will need to pick > something that people either know or can learn quickly. > > Oh, and be prepared for people to give you unsolicited opinions > about all sorts of things. :) > > All that being said, I say go for it. Find what works for you and > see what happens. Be sure to use #freeswitch-gui. If this really > takes off I'm sure that we could even start a mailing list for GUI > development. > Once the goals and features decided I think more ppl can join and work this out together. > Enjoy! > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/be210a39/attachment-0002.html From regs at kinetix.gr Fri Jun 5 01:35:52 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 05 Jun 2009 11:35:52 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> <4A2808A9.8070409@kinetix.gr> <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> Message-ID: <4A28D8E8.4040000@kinetix.gr> Anthony Minessale wrote: > FS uses async rtp timers so you may want to set rtp-timer-name=none in > the profile param to simulate asterisk conditions. I tried that - although I am not using rtp in my scenario - with the same results. > Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit > single cpu box because that was what was popular when it was designed > and the chance for race conditions is minimal because there is only 1 > cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic > difference. Yes I know that this machine is not well suited for today's test needs. But the issue occurs in every machine as long as it is pushed near (but not quite near) to its limits. I have the same odd durations using a 64 bit low end server. In this case I could achieve a better call/sec rate than that of the crappy PC but around 50-60 calls/sec the same problem showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the same thing happened at a higher rate. > > I will be happy to investigate this issue a bit if you'd like but i do > not have any box like you describe so if I can't find anything > you may have to lend us your lab. I would appreciate it if you did. After all there this might be a problem that has not surfaced yet but someday will as more and more production boxes start using FS. So it would be better to investigate it now. I don't think lending you access to my old P4 PC would help you very much :) If you have access to a normal 2-4 core system you can easily start raising the sipp parameters until it starts happening. However if you really think it is appropriate to use my test machines I'd be happy to grant access to our low-end Opteron machine (just send me a personal email). I cannot grant you access to larger systems because they are used in production. I used the embedded sipp scenarios : on the UAS side : sipp -i -mi -ci -mp 8000 -sn uas on the UAC side : sipp :5060 -s 44050505-i -mi -ci -r 70 -d 5000 -l 500 -m 2000 -sn uac The dialplan : If you need anything else from the config just notify me. In order to verify that at some point the calls start having a duration larger than the scenario's 5secs you can tcpdump on the sipp machine or turn on the cdrs logging (I know that it degrades performance, but as I said it is not a matter of when exactly it starts happening, it is a matter that it DOES start happening). > > > On Thu, Jun 4, 2009 at 12:47 PM, regs at kinetix.gr > > wrote: > > Michael Collins wrote: > > > > > > The dialplan : > > > > > > > > > > > > > > > > > expression="^.*$"> > > > > > > You forgot the parens around .* > > It should be expression="^(.*)$" if you plan to use $1 later in the > > extension... > > > > > > > > > > > data="absolute_codec_string=PCMA"/> > > > data="sofia/gateway/sipp01/$1"/> > > > > ... like here ^^^^^^^ > > :) > > -MC > > You are right! Although, I don't think that would change the outcome of > my test :) > > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From dujinfang at gmail.com Fri Jun 5 02:10:45 2009 From: dujinfang at gmail.com (seven) Date: Fri, 5 Jun 2009 17:10:45 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> Message-ID: <5AC703CD-2434-4F1A-AFF1-0D5DAFDAF800@gmail.com> On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > I'm using openzap analog with tone_detect, it works(conference not > tested). however, according to the asterisk book, Kewlstart can > detect the busy tone and disconnect the circuit. does anyone knows > how to configure kewlstart with freeswitch/openzap? guess we don't > need tone_detect then. > > Dujinfang, > > Your telco must support "kewlstart" signaling for this to be > effective. The telco probably calls it something different, like > "disconnect supervision" or "drop in loop current" or "battery > reversal" or something like that. In any case, if the signaling is > supported then you need to set up your zaptel.conf with the > appropriate signaling type, which is either fxoks or fxsks. (I can > never remember because zaptel does it backwards where if you have an > FXO port then it uses FXS signaling but if you have an FXS port it > uses FXO signaling. Stupidity, to be sure, so be aware of it.) > > Find the sample zaptel.conf that comes with the zaptel package and > search it for fxsks or fxoks and you'll see some notes on how to set > it up for your analog trunks. > > -MC Thank you for the detailed explain MC. so the ks means kewlstart, it already set, but no luck. anyway, the tone_detect works for me, less worry about that. Thanks again. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/5cd996f5/attachment-0002.html From wasim at convergence.pk Fri Jun 5 03:02:32 2009 From: wasim at convergence.pk (Wasim Baig) Date: Fri, 5 Jun 2009 16:02:32 +0600 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> References: <4A283DCC.5040701@gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: Just to chime in, perhaps GWT might be a good framework ... -wasim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/927e9833/attachment-0002.html From tayeb.meftah at gmail.com Fri Jun 5 03:16:13 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 05 Jun 2009 10:16:13 +0000 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: <4A28F06D.9020701@gmail.com> hello wasim, GWT will help me about Accessibility, is very very accessible tel me about how i can host GWT application thanks Wasim Baig wrote: > Just to chime in, perhaps GWT might be a good framework ... > > -wasim > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/95a19c31/attachment-0002.html From matthew.lockwood at gmail.com Fri Jun 5 03:30:51 2009 From: matthew.lockwood at gmail.com (Matthew Lockwood) Date: Fri, 5 Jun 2009 03:30:51 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> References: <4A283DCC.5040701@gmail.com> <87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: <415541b10906050330w35a1369al2d8fe075c38a59c0@mail.gmail.com> Thanks for all this. What I meant to respond with before I passed out asleep was: The framework doesn't matter that much. We've all got looped on this issue. Everything that is done has to add value to the end user. The framework is far down at the bottom of the list of things that provides value, but it's not something to be ignored. The vision I have for this is something that's so simple it lowers the barriers that would otherwise stop people from using FreeSwitch. Using some relatively unheard of framework is going to most certainly complicate things. Simple = good. And plus, on a side note if we throw out a whole bunch of frameworks and acronyms and make a big deal about the actual technology that powers the GUI (not that people even care most of the time), people will start to get more confused and it'll backfire. I'll be using .NET/Mono unless I can come up with an exceptionally good reason to use something else. I'm choosing this framework over everything else because it's what I know best and I'll be writing the code base. I've got years of experience writing code in C# and developing .NET web applications so it makes more sense than learning something new that will slow the development time and result in me producing poorer code. This isn't me being mercenary, but the GUI isn't likely to cross the million codeline barrier (even with everything implemented) and this is a framework I have a lot of experience with. I'm totally fine being the lone developer for now, and there are a lot of people with similar programming skillsets as mine so it's not like there will never be anybody else that'll ever contribute code. Personally, I think it's more important to have well written code that is rapidly developed than it is to have a shiny technology that adds no value. :-) Of course, the final product will be perfectly standards compliant and 100% accessible. I know this is important. I'm going to lay the framework issue to rest now. It'll be .NET/Mono unless there is some super-compelling reason to use something else. If for some reason there is such a reason not to use .NET/Mono, the second choice is PHP. The other thing is I'm pretty much going to develop upwards of 95% of the features in one go. Nobody wants an incomplete product that lacks necessary functionality, so from v1.0 it'll be pretty much feature complete. I'm developing this for use in my business, so I need it feature complete, and that's what the community will get too - a feature complete product. Hope you're happy having a fully fledged GUI! ;-) M On Fri, Jun 5, 2009 at 1:26 AM, seven wrote: > > On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: > > > > On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood < > matthew.lockwood at gmail.com> wrote: > >> That would involve me learning a totally new framework. It'll not the >> hardest code I'll ever write by far, so I'm okay coding it up on my own. >> However, I definitely need a lot of help from fabulous designers to actually >> make the interface pretty and useable. Plus, I'm only one person and will >> need a lot of feedback to create something that rocks - everybody has a >> different use case and I can't foresee how everybody will use it, so that >> kind of feedback will go into re-engineering it. >> > > If you guys are serious about this then I would like to make a few > suggestions that might be obvious but for the sake of the project we'll make > them explicitly obvious. > > First, before deciding what framework to use, it would be good to hold some > discussions about what the GUI actually needs to do: > What are the design goals? > Will it be just for setting up extensions and the dialplan? Or will it go > much farther than that? > Will you be using mod_xml_curl for everything? If so, what database(s) will > you support? > Are you going to have extra goodies like an IVR builder? > A 'visual voicemail' page? > A user portal? > Management interface to 'spy' on users? > A CDR/call accounting system? > FIFO and/or ACD queue management? > MOH and sound files management? > > > and conference management > > > > It's okay to start small and build your way out, but you need to know > before you start building what the grand scheme will be. The larger the > goals of the project, the narrower your choices for a framework that can do > it all. The simple fact of the matter is that if you want to use a MVC web > framework then you have a somewhat limited number of choices. You need a MVC > WF that fits your needs, which means it needs to be at least somewhat > flexible. If you want a pretty GUI then you need to decide if you want a > rich Internet application (RIA) front end like AIR, or do you want something > along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you > cross-browser widgets and tools. All of this on top of the fact that if you > want volunteers to assist you will need to pick something that people either > know or can learn quickly. > > Oh, and be prepared for people to give you unsolicited opinions about all > sorts of things. :) > > All that being said, I say go for it. Find what works for you and see what > happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure > that we could even start a mailing list for GUI development. > > > Once the goals and features decided I think more ppl can join and work this > out together. > > Enjoy! > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/bec4fff3/attachment-0002.html From d at d-man.org Fri Jun 5 03:35:34 2009 From: d at d-man.org (Darren Schreiber) Date: Fri, 5 Jun 2009 03:35:34 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> References: <4A283DCC.5040701@gmail.com> <4A284A90.70602@gmail.com><415541b10906041534p38c87428n158a4189806d8c61@mail.gmail.com><87f2f3b90906041540s4bb0f1d3heef619dd15c114bc@mail.gmail.com><415541b10906041542t46819552s531a54db86c58754@mail.gmail.com><389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org><415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com><415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com><9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com><415541b10906041852g5441267br157501bf7297b313@mail.gmail.com><87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: This is a really ironic post, Seven. :-) I agree with all your points. A while ago I started the TCAPI project to build a front-end for FreeSWITCH. I very quickly got inundated with debates about framework and language. These debates were initially appreciated but at some point we needed to decide & move on. The real work to be done was, as you point out, in design of the application business logic, interface and actually coding it up and putting it together. So we decided to go a bit radio silent and and focus on a few developers who were willing to build out the foundational pieces of the MVC architecture, and to let you create FreeSWITCH config files and general database and software modules with a set of standardized, simple to use libraries/APIs. Once we are done with that, the intention was to release it to those who wanted to help build the pieces related to modules in FreeSWITCH. That project is about 6 weeks from release into beta, give or take a few weeks (hey, it's software dev! heh who's ever on time?). So anyone who is on here reading this and might be interested in contributing code to an already very active FreeSWITCH GUI development project please feel free to contact me - we are now accepting serious developer inquiries. The project is in PHP and uses two pretty nifty frameworks (we, as you point out, couldn't find exactly what we were looking for, so we merged two libraries that fit the bill very nicely). It is database agnostic and is designed to work on Windows or Linux so don't let that be a barrier to participation. This will be an open source project for all, btw. I will be presenting on it at the upcoming ClueCon, warts and all, so you should go register and then you can participate in the demo/tutorial! :-) - Darren _____ From: seven [mailto:dujinfang at gmail.com] Sent: Friday, June 05, 2009 1:26 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] WikiPBX Installation On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood wrote: That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? and conference management It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited opinions about all sorts of things. :) All that being said, I say go for it. Find what works for you and see what happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure that we could even start a mailing list for GUI development. Once the goals and features decided I think more ppl can join and work this out together. Enjoy! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/9230ded1/attachment-0002.html From larclap at yahoo.com Fri Jun 5 06:00:44 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 5 Jun 2009 06:00:44 -0700 Subject: [Freeswitch-users] Missing lines copying data from console to vi In-Reply-To: References: <079f01c9e535$c1fed2e0$45fc78a0$@com> <08bc01c9e57e$5ef60030$1ce20090$@com> Message-ID: <09a501c9e5dd$a36e2e90$ea4a8bb0$@com> Thanks, Jim, that advice really helped in more ways than I asked for. Lars -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim Burke Sent: Thursday, June 04, 2009 6:52 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing lines copying data from console to vi After logging into my linux box using putty. I then change directory the the ~/freeswitch/bin directory and run ./fs_cli You can, but you don't need to start FS from the putty terminal. We run ours as a background process. On Fri, Jun 5, 2009 at 11:38 AM, Lars Zeb wrote: > Does this mean that I must start fs from the putty terminal, or can I attach > to an already running instance via putty? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim > Burke > Sent: Thursday, June 04, 2009 4:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Missing lines copying data from console to > vi > > Not that this helps your question directly. ?Using a putty terminal > from windows allows the data to be copied correctly from console > sessions. > > > > On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb wrote: >> I want to copy the results of a siptrace captured on the fs console to a >> file. The console is running on a Gnome terminal. I highlight the text I >> want to copy in the fs console, open a vi session in insert mode, and > paste >> the text. However, the text is not pasted as I copied it - it is missing >> characters/lines. >> >> >> >> I know I am doing something wrong. Is there another way to save siptraces > to >> a file? Redirection doesn't work. >> >> >> >> sofia profile internal siptrace on is the command I use. >> >> >> >> Thanks Lars >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Jim Burke > Director Evolutiontel. > http://www.evolutiontel.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From shoaib at breezecom.ae Thu Jun 4 22:35:13 2009 From: shoaib at breezecom.ae (Shoaib Khanzada) Date: Fri, 5 Jun 2009 11:35:13 +0600 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: <001b01c9e59f$68ffdfd0$3aff9f70$@ae> Hi FS-Users, I am having an strange problem of sessions in freeswitch. For every call Freeswitch is creating more than two sessions (in and out legs). For example, I have seen 260+ sessions for just 30+ calls whereas there should not be more than 60 sessions for just 30 calls. I have seen the same problem with 1.0.4pre8 and trunk version. I am using default values from switch.conf.xml for max_sessions (1000) and sessions_per_second(30). Freeswitch create many sessions only when there is good load on the system. It works fine with the steady load of 50-100 calls. However, if I give it 200+ calls at once then it breaks. Any suggestion? Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/ecc46a4a/attachment-0002.html From brian at freeswitch.org Fri Jun 5 07:07:13 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 09:07:13 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: <3wmhHdntwTK5.tvBsy3CM@smtp.gmail.com> References: <3wmhHdntwTK5.tvBsy3CM@smtp.gmail.com> Message-ID: DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > should open a jira on this? Did you guys need any more info? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/50d872ab/attachment-0002.html From anthony.minessale at gmail.com Fri Jun 5 07:08:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Jun 2009 09:08:09 -0500 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: <001b01c9e59f$68ffdfd0$3aff9f70$@ae> References: <001b01c9e59f$68ffdfd0$3aff9f70$@ae> Message-ID: <191c3a030906050708q7fc62c74j896b08f0454864d1@mail.gmail.com> No real suggestions because there is very little information here. We tend to not address load testing issues because most problems are caused by improper test environments and user error. My suggestion is to try it for real instead of in a test land. On Fri, Jun 5, 2009 at 12:35 AM, Shoaib Khanzada wrote: > Hi FS-Users, > > > > I am having an strange problem of sessions in freeswitch. > > > > For every call Freeswitch is creating more than two sessions (in and out > legs). For example, I have seen 260+ sessions for just 30+ calls whereas > there should not be more than 60 sessions for just 30 calls. > > > > I have seen the same problem with 1.0.4pre8 and trunk version. > > > > I am using default values from switch.conf.xml for max_sessions (1000) and > sessions_per_second(30). > > > > Freeswitch create many sessions only when there is good load on the system. > It works fine with the steady load of 50-100 calls. However, if I give it > 200+ calls at once then it breaks. > > > > Any suggestion? > > > > Shoaib > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/e1d3db74/attachment-0002.html From msc at freeswitch.org Fri Jun 5 09:27:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 5 Jun 2009 09:27:37 -0700 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: References: <4A283DCC.5040701@gmail.com> <415541b10906041542t46819552s531a54db86c58754@mail.gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> Message-ID: <87f2f3b90906050927t6022fab9ka1718e8cec98727c@mail.gmail.com> On Fri, Jun 5, 2009 at 3:35 AM, Darren Schreiber wrote: > This is a really ironic post, Seven. :-) I agree with all your points. > > A while ago I started the TCAPI project to build a front-end for > FreeSWITCH. I very quickly got inundated with debates about framework and > language. These debates were initially appreciated but at some point we > needed to decide & move on. The real work to be done was, as you point out, > in design of the application business logic, interface and actually coding > it up and putting it together. So we decided to go a bit radio silent and > and focus on a few developers who were willing to build out the foundational > pieces of the MVC architecture, and to let you create FreeSWITCH config > files and general database and software modules with a set of standardized, > simple to use libraries/APIs. Once we are done with that, the intention was > to release it to those who wanted to help build the pieces related to > modules in FreeSWITCH. That project is about 6 weeks from release into > beta, give or take a few weeks (hey, it's software dev! heh who's ever on > time?). > > I was wondering when you were gonna chime in on this subject! :D -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/6c223767/attachment-0002.html From marcus.frenkel at gmail.com Fri Jun 5 11:14:15 2009 From: marcus.frenkel at gmail.com (Marcus Frenkel) Date: Fri, 5 Jun 2009 20:14:15 +0200 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH Message-ID: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> Hi, I'm using GnuGK H323 gatekeeper. It has good performance and many features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch comparisons, but how about GnuGK vs FreeSwitch? The setup for which I'm asking is in a range of 200 concurrent calls. The points that I'm really interested for comparison are: 1) Proxy of RTP feature and it's stability 2) NAT support 3) Direct SQL AAA support (without the need of using RADIUS server) 4) Performance as an endpoint registrar 5) Rerouting to a second carrier on failed call Also, is the H323 library of FreeSWITCH based on h323plus/openh323? Marcus From anthony.minessale at gmail.com Fri Jun 5 12:47:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Jun 2009 14:47:45 -0500 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH In-Reply-To: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> References: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> Message-ID: <191c3a030906051247m542efe08l2011dd25c0e81c34@mail.gmail.com> the h323 is done with mod_opal On Fri, Jun 5, 2009 at 1:14 PM, Marcus Frenkel wrote: > Hi, > > I'm using GnuGK H323 gatekeeper. It has good performance and many > features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch > comparisons, but how about GnuGK vs FreeSwitch? > > The setup for which I'm asking is in a range of 200 concurrent calls. > > The points that I'm really interested for comparison are: > 1) Proxy of RTP feature and it's stability > 2) NAT support > 3) Direct SQL AAA support (without the need of using RADIUS server) > 4) Performance as an endpoint registrar > 5) Rerouting to a second carrier on failed call > > Also, is the H323 library of FreeSWITCH based on h323plus/openh323? > > Marcus > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/6f29d867/attachment-0002.html From edpimentl at gmail.com Fri Jun 5 13:11:28 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 5 Jun 2009 16:11:28 -0400 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH In-Reply-To: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> References: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> Message-ID: <9dc4a1670906051311m7bf6a7acw9ae9d8f327ef6d19@mail.gmail.com> Is this a joke or do want to save the question for April Fools? -E On Fri, Jun 5, 2009 at 2:14 PM, Marcus Frenkel wrote: > Hi, > > I'm using GnuGK H323 gatekeeper. It has good performance and many > features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch > comparisons, but how about GnuGK vs FreeSwitch? > > The setup for which I'm asking is in a range of 200 concurrent calls. > > The points that I'm really interested for comparison are: > 1) Proxy of RTP feature and it's stability > 2) NAT support > 3) Direct SQL AAA support (without the need of using RADIUS server) > 4) Performance as an endpoint registrar > 5) Rerouting to a second carrier on failed call > > Also, is the H323 library of FreeSWITCH based on h323plus/openh323? > > Marcus > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/94dd9710/attachment-0002.html From skhanzada at gmail.com Fri Jun 5 13:09:54 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 02:09:54 +0600 Subject: [Freeswitch-users] Reuse ODBC connection in javascript Message-ID: <8eace5520906051309l4916a1d9p3c95497df0f0b17f@mail.gmail.com> Hi, I want to reuse database connection (mysql) in one of my javascript which is executed on each call. Following is how I am creating ODBC connection. Line1) var db = new ODBC(DSN, DB_USER, DB_PASS); Line2) db.connect(); My first question is, where does it create a database connection on line1 or on line2? Secondly, how can I reuse this connection? so that it is not created for each call and I just use a previously created db object in my routing script. My Objective is: 1) Create ODBC connection on freeswitch startup (could be a database connection pool) 2) Reuse the connection on each call 3) Close database connection on freeswitch shutdown (or on some other event) Any help would be highly appreciated. Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/431b0b94/attachment-0002.html From skhanzada at gmail.com Fri Jun 5 13:10:21 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 02:10:21 +0600 Subject: [Freeswitch-users] Using setGlobalVar and getGlobalVar Message-ID: <8eace5520906051310i79f3d037lcb1765ef2ac158b4@mail.gmail.com> Hi, How can I use setGlobalVar and getGlobalVar in my javascript to store a ODBC connection? I want to set an ODBC database connection object globally so that I can access it from anywhere. This connection will be used for read-only so no threading issues. Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/764e7778/attachment-0002.html From skhanzada at gmail.com Fri Jun 5 13:11:15 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 02:11:15 +0600 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: <8eace5520906051311u19d36449scd447b05975b605d@mail.gmail.com> Thanks for the reply?. I was not running it for testing purpose because I?ve completed all testing successfully. We are running freeswitch in a voip carrier grade environment. It works perfectly alright in off peak hours when we have low calls ratio around 100-150 concurrent calls. However, in peak hours this goes beyond 200 calls and on that point freeswitch start creating many sessions for each calls. I have seen the sessions using ?status? command and calls using ?show calls count?. We are using javascript to select the route from the mysql database for each call. Could it be because script is taking longer than expected amount of time to retrieve a route? and freeswitch continuously keep creating sessions for incoming calls. That?s why I see low no of connected calls (if ?show calls count? only display the connected calls) whereas sessions are continuously being created by freeswitch as it is receiving many calls. If above text confuses you, nevermind just answer the following questions. 1) Does ?show calls count? display the connected calls only? 2) When freeswitch create session instances? Before bridge or after bridge? Or one before bridge and one after bridge? Thanks, Shoaib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/48b934e6/attachment-0002.html From mike at jerris.com Fri Jun 5 13:23:23 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Jun 2009 16:23:23 -0400 Subject: [Freeswitch-users] Using setGlobalVar and getGlobalVar In-Reply-To: <8eace5520906051310i79f3d037lcb1765ef2ac158b4@mail.gmail.com> References: <8eace5520906051310i79f3d037lcb1765ef2ac158b4@mail.gmail.com> Message-ID: On Jun 5, 2009, at 4:10 PM, Shoaib Khanzada wrote: > Hi, > > How can I use setGlobalVar and getGlobalVar in my javascript to > store a ODBC connection? > > I want to set an ODBC database connection object globally so that I > can access it from anywhere. This connection will be used for read- > only so no threading issues. > no, those are for strings only. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/e0142ff9/attachment-0002.html From mike at jerris.com Fri Jun 5 13:25:38 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Jun 2009 16:25:38 -0400 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: <8eace5520906051311u19d36449scd447b05975b605d@mail.gmail.com> References: <8eace5520906051311u19d36449scd447b05975b605d@mail.gmail.com> Message-ID: <40DD2D82-68BF-4898-8233-636D6EBB2E22@jerris.com> On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: > Thanks for the reply?. > > I was not running it for testing purpose because I?ve completed all > testing successfully. > > We are running freeswitch in a voip carrier grade environment. It > works perfectly alright in off peak hours when we have low calls > ratio around 100-150 concurrent calls. However, in peak hours this > goes beyond 200 calls and on that point freeswitch start creating > many sessions for each calls. I have seen the sessions using > ?status? command and calls using ?show calls count?. > > We are using javascript to select the route from the mysql database > for each call. Could it be because script is taking longer than > expected amount of time to retrieve a route? and freeswitch > continuously keep creating sessions for incoming calls. That?s why I > see low no of connected calls (if ?show calls count? only display > the connected calls) whereas sessions are continuously being created > by freeswitch as it is receiving many calls. > > If above text confuses you, nevermind just answer the following > questions. > > 1) Does ?show calls count? display the connected calls only? Only bridged calls (2 sessions) > 2) When freeswitch create session instances? Before bridge or > after bridge? Or one before bridge and one after bridge? It creates a session when it gets an incomming call and creates one for each outgoing call, unrelated to bridging. > > Thanks, > > Shoaib How are you doing the bridge in your script? Are you setting a var then dropping out of the js to do the bridge? Can you post your js file? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/0b060607/attachment-0002.html From marcus.frenkel at gmail.com Fri Jun 5 14:55:14 2009 From: marcus.frenkel at gmail.com (Marcus Frenkel) Date: Fri, 5 Jun 2009 23:55:14 +0200 Subject: [Freeswitch-users] GnuGK vs FreeSWITCH In-Reply-To: <9dc4a1670906051311m7bf6a7acw9ae9d8f327ef6d19@mail.gmail.com> References: <1ded54630906051114u5500de0ds7d94b840fd27e77a@mail.gmail.com> <9dc4a1670906051311m7bf6a7acw9ae9d8f327ef6d19@mail.gmail.com> Message-ID: <1ded54630906051455r60ed2194he8b1ad22f64d4f63@mail.gmail.com> Can you explain your friendly advice? Some parts can be compared On Fri, Jun 5, 2009 at 10:11 PM, EdPimentl wrote: > Is this a joke or do want to save the question for April Fools? > -E > > > On Fri, Jun 5, 2009 at 2:14 PM, Marcus Frenkel > wrote: >> >> Hi, >> >> I'm using GnuGK H323 gatekeeper. It has good performance and many >> features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch >> comparisons, but how about GnuGK vs FreeSwitch? >> >> The setup for which I'm asking is in a range of 200 concurrent calls. >> >> The points that I'm really interested for comparison are: >> 1) Proxy of RTP feature and it's stability >> 2) NAT support >> 3) Direct SQL AAA support (without the need of using RADIUS server) >> 4) Performance as an endpoint registrar >> 5) Rerouting to a second carrier on failed call >> >> Also, is the H323 library of FreeSWITCH based on h323plus/openh323? >> >> Marcus >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jim at evolutiontel.net Fri Jun 5 16:25:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Sat, 6 Jun 2009 09:25:39 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. Message-ID: Yup sure did, same result :( - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Brian West Date: 05/06/2009 14:10 DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > should open a jira on this? Did you guys need any more info? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Jun 5 16:53:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 5 Jun 2009 18:53:02 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: References: Message-ID: <191c3a030906051653nd8c56d4wb50a6ab64e1a259f@mail.gmail.com> it is using G729a which is not the correct RFC value that goes with payload 18 so it's moving it to 96 as if it's some other codec. the only thing you can do is get them to stop using invalid data in their sdp or hack it to replace "G729a" with "G729 " before it's too late. On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/777cc863/attachment-0002.html From mrene_lists at avgs.ca Fri Jun 5 16:54:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 5 Jun 2009 19:54:48 -0400 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: <191c3a030906051653nd8c56d4wb50a6ab64e1a259f@mail.gmail.com> References: <191c3a030906051653nd8c56d4wb50a6ab64e1a259f@mail.gmail.com> Message-ID: <68EDAD18-27B1-4BB5-86E1-573A55F0D72C@avgs.ca> Linksys still uses G729a in their sdp, but you can change it in the admin panel (if thats what you have) Math On 5-Jun-09, at 7:53 PM, Anthony Minessale wrote: > it is using G729a which is not the correct RFC value that goes with > payload 18 so it's moving it to 96 as if it's some other > codec. > > the only thing you can do is get them to stop using invalid data in > their sdp > or hack it to replace "G729a" with "G729 " before it's too late. > > > On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke > wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/bf1ba320/attachment-0002.html From jim at evolutiontel.net Fri Jun 5 17:00:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Sat, 6 Jun 2009 10:00:39 +1000 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: Could it be that you are getting several INVITE messages per call due to 100 Trying message is not being sent until after the route is selected from your java routing script? You might be able to send a 100 trying from your dialplan or script. Keep in mind that this message stops timers on the originating side. - original message - Subject: Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call From: Michael Jerris Date: 05/06/2009 20:27 On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: > Thanks for the reply?. > > I was not running it for testing purpose because I?ve completed all > testing successfully. > > We are running freeswitch in a voip carrier grade environment. It > works perfectly alright in off peak hours when we have low calls > ratio around 100-150 concurrent calls. However, in peak hours this > goes beyond 200 calls and on that point freeswitch start creating > many sessions for each calls. I have seen the sessions using > ?status? command and calls using ?show calls count?. > > We are using javascript to select the route from the mysql database > for each call. Could it be because script is taking longer than > expected amount of time to retrieve a route? and freeswitch > continuously keep creating sessions for incoming calls. That?s why I > see low no of connected calls (if ?show calls count? only display > the connected calls) whereas sessions are continuously being created > by freeswitch as it is receiving many calls. > > If above text confuses you, nevermind just answer the following > questions. > > 1) Does ?show calls count? display the connected calls only? Only bridged calls (2 sessions) > 2) When freeswitch create session instances? Before bridge or > after bridge? Or one before bridge and one after bridge? It creates a session when it gets an incomming call and creates one for each outgoing call, unrelated to bridging. > > Thanks, > > Shoaib How are you doing the bridge in your script? Are you setting a var then dropping out of the js to do the bridge? Can you post your js file? Mike _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Fri Jun 5 19:11:58 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 6 Jun 2009 10:11:58 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> Message-ID: <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: > > > On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > I'm using openzap analog with tone_detect, it works(conference not > tested). however, according to the asterisk book, Kewlstart can > detect the busy tone and disconnect the circuit. does anyone knows > how to configure kewlstart with freeswitch/openzap? guess we don't > need tone_detect then. > > Dujinfang, > > Your telco must support "kewlstart" signaling for this to be > effective. The telco probably calls it something different, like > "disconnect supervision" or "drop in loop current" or "battery > reversal" or something like that. In any case, if the signaling is > supported then you need to set up your zaptel.conf with the > appropriate signaling type, which is either fxoks or fxsks. (I can > never remember because zaptel does it backwards where if you have an > FXO port then it uses FXS signaling but if you have an FXS port it > uses FXO signaling. Stupidity, to be sure, so be aware of it.) > 1) Don't know why but the similar zaptel.conf works on asterisk. I guess tone_detect in FS is equivalent to busydetect=yes in Asterisk(zapata.conf) . zaptel.conf # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" fxoks=5 fxoks=6 fxoks=7 fxoks=8 # Global data loadzone = us defaultzone = us zapata.conf usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes I agree the FXO and FXS signaling is weird, why not they just match the care name and reverse that internally? 2) Another essue is if I dial out from a FXO port from a local extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo on asterisk. the zt.conf as below and I tried to change the echo_cancel_level to 32 or 128 got no much difference. Is there any equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? can I set busydetect and echocancelwhenbridged and other options like this ? [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 > Find the sample zaptel.conf that comes with the zaptel package and > search it for fxsks or fxoks and you'll see some notes on how to set > it up for your analog trunks. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/d6b9464d/attachment-0002.html From gcd at i.ph Fri Jun 5 19:24:09 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 6 Jun 2009 10:24:09 +0800 Subject: [Freeswitch-users] Reducing record_session load Message-ID: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> we experience some latency in the recording files even with PCMU-PCMU session to a stereo WAV file. i want to reduce the CPU load hoping to reduce this problem. would it help if do the ff? 1. save it in PCMU file. i can use sox at the end of the shift. 2. record in mono. does it help? 3. will record_session work w/ proxy_media=true? tks for your help. -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/88e9f6f4/attachment-0002.html From jim at evolutiontel.net Fri Jun 5 19:25:32 2009 From: jim at evolutiontel.net (Jim Burke) Date: Sat, 6 Jun 2009 12:25:32 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. Message-ID: Yes, the terminator is Linksys so will change it and test. Noticed there is a list of mime types associated with FS and G729a was not listed, does this have anything to do with the root cause? - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Mathieu Rene Date: 05/06/2009 23:56 Linksys still uses G729a in their sdp, but you can change it in the admin panel (if thats what you have) Math On 5-Jun-09, at 7:53 PM, Anthony Minessale wrote: > it is using G729a which is not the correct RFC value that goes with > payload 18 so it's moving it to 96 as if it's some other > codec. > > the only thing you can do is get them to stop using invalid data in > their sdp > or hack it to replace "G729a" with "G729 " before it's too late. > > > On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke > wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Fri Jun 5 19:32:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 21:32:01 -0500 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> Message-ID: On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote: > we experience some latency in the recording files even with PCMU- > PCMU session to a stereo WAV file. i want to reduce the CPU load > hoping to reduce this problem. would it help if do the ff? > 1. save it in PCMU file. i can use sox at the end of the shift. You shouldn't be experiencing this at all... how many are you doing at once? > 2. record in mono. does it help? No. > 3. will record_session work w/ proxy_media=true? Nope. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/fa3c9cdb/attachment-0002.html From brian at freeswitch.org Fri Jun 5 19:32:37 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 21:32:37 -0500 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: References: Message-ID: <51DECC6F-E822-4459-BF47-970576041825@freeswitch.org> G729a is 100% INVALID in the sdp on codec 18. /b On Jun 5, 2009, at 9:25 PM, Jim Burke wrote: > Noticed there is a list of mime types associated with FS and G729a > was not listed, does this have anything to do with the root cause? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/95ea73ad/attachment-0002.html From gcd at i.ph Fri Jun 5 19:58:05 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 6 Jun 2009 10:58:05 +0800 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> Message-ID: <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> there 10 client seats so at max. 10 simultaneous calls. however, the number of clients may be increased. -nandy On Sat, Jun 6, 2009 at 10:32 AM, Brian West wrote: > > On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote: > > we experience some latency in the recording files even with PCMU-PCMU > session to a stereo WAV file. i want to reduce the CPU load hoping to reduce > this problem. would it help if do the ff? > 1. save it in PCMU file. i can use sox at the end of the shift. > > > You shouldn't be experiencing this at all... how many are you doing at > once? > > 2. record in mono. does it help? > > > No. > > 3. will record_session work w/ proxy_media=true? > > > Nope. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/e0e8af65/attachment-0002.html From brian at freeswitch.org Fri Jun 5 20:02:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 22:02:53 -0500 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> Message-ID: <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> You shouldn't be having problems... what version are you using? /b On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: > there 10 client seats so at max. 10 simultaneous calls. however, the > number of clients may be increased. > -nandy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/39558777/attachment-0002.html From gcd at i.ph Fri Jun 5 20:03:20 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 6 Jun 2009 11:03:20 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> Message-ID: <7d0bfd8c0906052003w40a8fff4s8644cb9773e346b5@mail.gmail.com> dujinfang, hv u tried OSLEC? it's really reduced echo even on the cheapy X100P card on *. oslec works w/ FS, too. -nandy On Sat, Jun 6, 2009 at 10:11 AM, dujinfang wrote: > > On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: > > > > On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: > >> I'm using openzap analog with tone_detect, it works(conference not >> tested). however, according to the asterisk book, Kewlstart can detect the >> busy tone and disconnect the circuit. does anyone knows how to configure >> kewlstart with freeswitch/openzap? guess we don't need tone_detect then. >> > > Dujinfang, > > Your telco must support "kewlstart" signaling for this to be effective. The > telco probably calls it something different, like "disconnect supervision" > or "drop in loop current" or "battery reversal" or something like that. In > any case, if the signaling is supported then you need to set up your > zaptel.conf with the appropriate signaling type, which is either fxoks or > fxsks. (I can never remember because zaptel does it backwards where if you > have an FXO port then it uses FXS signaling but if you have an FXS port it > uses FXO signaling. Stupidity, to be sure, so be aware of it.) > > > 1) Don't know why but the similar zaptel.conf works on asterisk. I guess > tone_detect in FS is equivalent to busydetect=yes in > Asterisk(zapata.conf) . > > > zaptel.conf > # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) > fxsks=1 > fxsks=2 > fxsks=3 > fxsks=4 > > # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" > fxoks=5 > fxoks=6 > fxoks=7 > fxoks=8 > > # Global data > > loadzone = us > defaultzone = us > > > zapata.conf > > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=no > ;echotraining=800 > rxgain=0.0 > txgain=0.0 > group=0 > callgroup=1 > pickupgroup=1 > immediate=no > busydetect=yes > > I agree the FXO and FXS signaling is weird, why not they just match the > care name and reverse that internally? > > 2) Another essue is if I dial out from a FXO port from a local > extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo > on asterisk. the zt.conf as below and I tried to change the > echo_cancel_level to 32 or 128 got no much difference. Is there any > equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? > can I set busydetect and echocancelwhenbridged and other options like > this ? > > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > echo_cancel_level => 64 > > > Find the sample zaptel.conf that comes with the zaptel package and search > it for fxsks or fxoks and you'll see some notes on how to set it up for your > analog trunks. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/de5569e5/attachment-0002.html From mike at jerris.com Fri Jun 5 20:17:03 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 5 Jun 2009 23:17:03 -0400 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: References: Message-ID: We already send a 100 before the call even hits the dialplan. Mike On Jun 5, 2009, at 8:00 PM, "Jim Burke" wrote: > Could it be that you are getting several INVITE messages per call > due to 100 Trying message is not being sent until after the route is > selected from your java routing script? > > You might be able to send a 100 trying from your dialplan or script. > Keep in mind that this message stops timers on the originating side. > > - original message - > Subject: Re: [Freeswitch-users] Freeswitch creating more then two > sessions for one call > From: Michael Jerris > Date: 05/06/2009 20:27 > > > On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: > >> Thanks for the reply?. >> >> I was not running it for testing purpose because I?ve completed all >> testing successfully. >> >> We are running freeswitch in a voip carrier grade environment. It >> works perfectly alright in off peak hours when we have low calls >> ratio around 100-150 concurrent calls. However, in peak hours this >> goes beyond 200 calls and on that point freeswitch start creating >> many sessions for each calls. I have seen the sessions using >> ?status? command and calls using ?show calls count?. >> >> We are using javascript to select the route from the mysql database >> for each call. Could it be because script is taking longer than >> expected amount of time to retrieve a route? and freeswitch >> continuously keep creating sessions for incoming calls. That?s why I >> see low no of connected calls (if ?show calls count? only display >> the connected calls) whereas sessions are continuously being created >> by freeswitch as it is receiving many calls. >> >> If above text confuses you, nevermind just answer the following >> questions. >> >> 1) Does ?show calls count? display the connected calls only? > > Only bridged calls (2 sessions) > >> 2) When freeswitch create session instances? Before bridge or >> after bridge? Or one before bridge and one after bridge? > > It creates a session when it gets an incomming call and creates one > for each outgoing call, unrelated to bridging. > >> >> Thanks, >> >> Shoaib > > How are you doing the bridge in your script? Are you setting a var > then dropping out of the js to do the bridge? Can you post your js > file? > > Mike > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From klaus.teller at gmx.net Fri Jun 5 20:34:26 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sat, 06 Jun 2009 05:34:26 +0200 Subject: [Freeswitch-users] How to reject a call without answering Message-ID: <20090606033426.72340@gmx.net> Hi, Going through the socket api, how can i reject a call without having to answer it first? I tried sending a hangup command with cause set either to NO_ANSWER or NORMAL_CLEARING. In both cases, Freeswitch does create another socket to deliver the very same call. More precisely, when a call comes in i send a connect command. Then after some few seconds, i then send the following hangup command: SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f call-command: hangup hangup-cause: NO_ANSWER Thanks for any feedback. Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From brian at freeswitch.org Fri Jun 5 20:45:29 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 5 Jun 2009 22:45:29 -0500 Subject: [Freeswitch-users] How to reject a call without answering In-Reply-To: <20090606033426.72340@gmx.net> References: <20090606033426.72340@gmx.net> Message-ID: <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> Try the respond app. /b On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: > Hi, > > Going through the socket api, how can i reject a call without having > to answer it first? > > I tried sending a hangup command with cause set either to NO_ANSWER > or NORMAL_CLEARING. In both cases, Freeswitch does create another > socket to deliver the very same call. > > More precisely, when a call comes in i send a connect command. Then > after some few seconds, i then send the following hangup command: > > SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f > call-command: hangup > hangup-cause: NO_ANSWER > > Thanks for any feedback. > > Klaus. > > -- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090605/055745ec/attachment-0002.html From klaus.teller at gmx.net Fri Jun 5 21:28:46 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sat, 06 Jun 2009 06:28:46 +0200 Subject: [Freeswitch-users] How to reject a call without answering In-Reply-To: <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> References: <20090606033426.72340@gmx.net> <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> Message-ID: <20090606042846.72320@gmx.net> It doesn't seem to work. I tried the following: api respond 9015430e-82cf-418c-bf4c-f3ac6e85caf2 503 SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 call-command: execute execute-app-name: respond execute-app-arg: 503 Is one of these what you meant? Klaus. -------- Original-Nachricht -------- > Datum: Fri, 5 Jun 2009 22:45:29 -0500 > Von: Brian West > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] How to reject a call without answering > Try the respond app. > > /b > > On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: > > > Hi, > > > > Going through the socket api, how can i reject a call without having > > to answer it first? > > > > I tried sending a hangup command with cause set either to NO_ANSWER > > or NORMAL_CLEARING. In both cases, Freeswitch does create another > > socket to deliver the very same call. > > > > More precisely, when a call comes in i send a connect command. Then > > after some few seconds, i then send the following hangup command: > > > > SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f > > call-command: hangup > > hangup-cause: NO_ANSWER > > > > Thanks for any feedback. > > > > Klaus. > > > > -- > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > -- GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 Euro/mtl.! http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a From jmesquita at gmail.com Fri Jun 5 21:47:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 6 Jun 2009 01:47:50 -0300 Subject: [Freeswitch-users] WikiPBX Installation In-Reply-To: <87f2f3b90906050927t6022fab9ka1718e8cec98727c@mail.gmail.com> References: <4A283DCC.5040701@gmail.com> <389C3811-3F43-44AD-BD24-DFF032291CE7@freeswitch.org> <415541b10906041548v3caed2e1r58775b9cba5cb440@mail.gmail.com> <415541b10906041613h2363c42eweaae396095a50d22@mail.gmail.com> <9dc4a1670906041847l516ae0d1lb089c79402372125@mail.gmail.com> <415541b10906041852g5441267br157501bf7297b313@mail.gmail.com> <87f2f3b90906042236o78ba5071l66436e5d323f3ce8@mail.gmail.com> <74AC21BC-25F5-407D-99A6-BC8395534581@gmail.com> <87f2f3b90906050927t6022fab9ka1718e8cec98727c@mail.gmail.com> Message-ID: <5a8712120906052147v45962d5r6539d40b153e89e7@mail.gmail.com> Hail to someone who has actually done something! :-) Darren, I am excited to started working on that with you all. Maybe adding mod_khomp support to it whenever I have some actual working code. jmesquita On Fri, Jun 5, 2009 at 1:27 PM, Michael Collins wrote: > > > On Fri, Jun 5, 2009 at 3:35 AM, Darren Schreiber wrote: > >> This is a really ironic post, Seven. :-) I agree with all your points. >> >> A while ago I started the TCAPI project to build a front-end for >> FreeSWITCH. I very quickly got inundated with debates about framework and >> language. These debates were initially appreciated but at some point we >> needed to decide & move on. The real work to be done was, as you point out, >> in design of the application business logic, interface and actually coding >> it up and putting it together. So we decided to go a bit radio silent and >> and focus on a few developers who were willing to build out the foundational >> pieces of the MVC architecture, and to let you create FreeSWITCH config >> files and general database and software modules with a set of standardized, >> simple to use libraries/APIs. Once we are done with that, the intention was >> to release it to those who wanted to help build the pieces related to >> modules in FreeSWITCH. That project is about 6 weeks from release into >> beta, give or take a few weeks (hey, it's software dev! heh who's ever on >> time?). >> >> > > I was wondering when you were gonna chime in on this subject! :D > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/e792228f/attachment-0002.html From mitul at enterux.com Fri Jun 5 22:23:30 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 6 Jun 2009 10:53:30 +0530 Subject: [Freeswitch-users] (no subject) Message-ID: <182EFB0F-8623-4A1B-A457-D171EC805E9F@enterux.com> Ttrfrtttgteruoywtklou Regards,juuyuuu Mitul Limbani, Founder & CEO, iuiiiiokljkkllllllllllllmmmmnnvvfcxzasqwwhjhyljljjifkkkljjyjjjkkjllgjjggllyjkljkokjkjjjujjjjjkmktdswwdsflyjhhhhhhbhh mmmmmlkkkjjjhhhjykvytyyp Enterux Solutions Pvt Ltd,bu. B. P The Enterprise Linux Company(r), http://www.enterux.com/i Pio From gmaruzz at celliax.org Fri Jun 5 23:23:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 6 Jun 2009 08:23:26 +0200 Subject: [Freeswitch-users] (no subject) In-Reply-To: <182EFB0F-8623-4A1B-A457-D171EC805E9F@enterux.com> References: <182EFB0F-8623-4A1B-A457-D171EC805E9F@enterux.com> Message-ID: <7b197bef0906052323m56ba507y75829094293f4771@mail.gmail.com> I agree! Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sat, Jun 6, 2009 at 7:23 AM, Mitul Limbani wrote: > Ttrfrtttgteruoywtklou > > Regards,juuyuuu > Mitul Limbani, > Founder & > CEO, > iuiiiiokljkkllllllllllllmmmmnnvvfcxzasqwwhjhyljljjifkkkljjyjjjkkjllgjjggllyjkljkokjkjjjujjjjjkmktdswwdsflyjhhhhhhbhh > mmmmmlkkkjjjhhhjykvytyyp > Enterux Solutions Pvt Ltd,bu. B. ?P > The Enterprise Linux Company(r), > http://www.enterux.com/i > Pio > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From skhanzada at gmail.com Sat Jun 6 00:53:59 2009 From: skhanzada at gmail.com (Shoaib Khanzada) Date: Sat, 6 Jun 2009 13:53:59 +0600 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call Message-ID: <8eace5520906060053s70ecd641y269834ec23d045b7@mail.gmail.com> Following is the js i am using to select route and bridge the call.... use("ODBC"); var DSN = "myodbc"; var DB_USER = "neo"; var DB_PASS = "....."; var sql; var prefix1; var ip1; var no2; var bridge_str; if(argv[0]=="") exit(); if(argv[1]=="") exit(); ip1 =argv[0].split("@")[1]; var no1=argv[1]; var sql = "SELECT id,prefix,name FROM internal_auth where substring('"+no1+"',1,length(prefix))=prefix and ip='" + ip1 +"' and active=1;"; var authName=""; var db = new ODBC(DSN, DB_USER, DB_PASS); db.connect(); db.query(sql); if(db.nextRow()) { var row = db.getData(); if (row["id"] == "" ) { console_log("Err", "auth failed(0001)\t" + no1 + "\targ[0]=>" + argv[0] + "\targ[1]=>" + argv[1] + "\n"); db.close(); exit(); } prefix1=row["prefix"]; authName=row["name"]; }else{ console_log("Err", "auth failed(0001)\t" + no1 + "\targ[0]=>" + argv[0] + "\targ[1]=>" + argv[1] + "\n"); db.close(); exit(); } no2=no1.substring(prefix1.length, no1.length); sql = "SELECT ec.* FROM auth_routes_carriers arc,external_carriers ec, codes c where arc.external_carriers_id = ec.id and arc.code_id = c.id and substring('"+no2+"',1,length(c.code))=c.code and ec.active=1 order by arc.priority asc"; db.connect(); db.query(sql); var carriersName=""; bridge_str=''; while(db.nextRow()) { var row = db.getData(); if (row["id"] != "" ) { bridge_str = row["gateway"]+row["prefix"]+no2+"\@"+row["ip"]; carriersName = row["name"]; break; } } db.close(); if(bridge_str != ''){ if(session.ready()) session.execute("bridge", bridge_str); } else { console_log("Err", "external carrier not found.\t" + no2 + "\targ[0]=>" + argv[0] + "\targ[1]=>" + argv[1] + "\n"); exit(); } exit(); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/f73b2bae/attachment-0002.html From rupa at rupa.com Sat Jun 6 06:26:05 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 6 Jun 2009 08:26:05 -0500 Subject: [Freeswitch-users] Freeswitch creating more then two sessions for one call In-Reply-To: <8eace5520906060053s70ecd641y269834ec23d045b7@mail.gmail.com> References: <8eace5520906060053s70ecd641y269834ec23d045b7@mail.gmail.com> Message-ID: Some thoughts: 1) look at mod_lcr 2) comments on script below... On Sat, Jun 6, 2009 at 2:53 AM, Shoaib Khanzada wrote: > Following is the js i am using to select route and bridge the call.... > > > use("ODBC"); > > var DSN = "myodbc"; > var DB_USER = "neo"; > var DB_PASS = "....."; > > [...] > var db = new ODBC(DSN, DB_USER, DB_PASS); > > db.connect(); > > [...] > > sql = "SELECT ec.* FROM auth_routes_carriers arc,external_carriers ec, > codes c where arc.external_carriers_id = ec.id and arc.code_id = c.id and > substring('"+no2+"',1,length(c.code))=c.code and ec.active=1 order by > arc.priority asc"; > > db.connect(); > no need to reconnect > > db.query(sql); > > [...] > > if(bridge_str != ''){ > if(session.ready()) session.execute("bridge", bridge_str); > Don't actually execute the bridge from javascript. Instead, set the bridge_str to a channel var and then do the bridge from the dialplan using that bridge_str. This way you don't have a javascript interpreter lying around for the duration of the call. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/868ef27d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Jun 6 10:39:13 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 6 Jun 2009 18:39:13 +0100 Subject: [Freeswitch-users] Problems subscribing to outbound socket events Message-ID: I've put some c++ test code together to let the outbound socket control the call, all works as expected, apart from the event subscription Sending myevents\n\n gives the channel events However sending event text all\n\n doesn't give me any events apart from the channel events. Anyone care to suggest what I might be doing wrong? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090606/b9e06398/attachment-0002.html From gcd at i.ph Sat Jun 6 16:46:14 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 7 Jun 2009 07:46:14 +0800 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> Message-ID: <7d0bfd8c0906061646p4e5fd28eh741d6a57064de32e@mail.gmail.com> i'm using version build 13245M on an Intel D945GCLF2 Atom Dual-core mobo w/ 2GB ram. -nandy On Sat, Jun 6, 2009 at 11:02 AM, Brian West wrote: > You shouldn't be having problems... what version are you using? > /b > > On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: > > there 10 client seats so at max. 10 simultaneous calls. however, the number > of clients may be increased. > -nandy > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/42a56f51/attachment-0002.html From gerry at pstn2.net Sat Jun 6 21:25:09 2009 From: gerry at pstn2.net (Gerry Hull) Date: Sun, 7 Jun 2009 00:25:09 -0400 Subject: [Freeswitch-users] I need a favor... Message-ID: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> OK, thanks to help on the list have my very cool FreeSwitch app running... Gotta love FS once you get over the learning hump! So, I build FS and got everything running smoothly on my Wndows development box. Great. Then I went to deploy it on a production server. As I figured, no copy-and-run here. I tried building the setup project but it's just not happening for me! Can someone out there build me the Windows MSI for build 13496 or later and provide a link to it? I'm in a bind here to get this up and running. If I can pry a few bux out of the boss, I hope to be a ClueCon and describe to application we have built with FreeSwitch. Regards, Gerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/200553b8/attachment-0002.html From tayeb.meftah at gmail.com Sun Jun 7 01:08:32 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 07 Jun 2009 08:08:32 +0000 Subject: [Freeswitch-users] I need a favor... In-Reply-To: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> References: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> Message-ID: <4A2B7580.9020909@gmail.com> hello, welcome, i'm able to build a Installer for your Freeswitch please: i have a very small Internet connection (128KBPS) and FS Setup syse is 40MB or +... i give you only the setup project and you compile it... ok? thanks Gerry Hull wrote: > OK, thanks to help on the list have my very cool FreeSwitch app > running... Gotta love FS once you get over the learning hump! > > So, I build FS and got everything running smoothly on my Wndows > development box. Great. Then I went to deploy it on a production > server. As I figured, no copy-and-run here. > > I tried building the setup project but it's just not happening for me! > > Can someone out there build me the Windows MSI for build 13496 or > later and provide a link to it? I'm in a bind here to get this up > and running. > > If I can pry a few bux out of the boss, I hope to be a ClueCon and > describe to application we have built with FreeSwitch. > > Regards, > > Gerry > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/ff5419ad/attachment-0002.html From anthony.minessale at gmail.com Sun Jun 7 07:32:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 7 Jun 2009 09:32:20 -0500 Subject: [Freeswitch-users] Reducing record_session load In-Reply-To: <7d0bfd8c0906061646p4e5fd28eh741d6a57064de32e@mail.gmail.com> References: <7d0bfd8c0906051924h76a3a8dds326ea518b2547818@mail.gmail.com> <7d0bfd8c0906051958g7a6e87d0wb5b72530bbe6b714@mail.gmail.com> <445E63EE-CA85-42BD-A0E2-9699B469F751@freeswitch.org> <7d0bfd8c0906061646p4e5fd28eh741d6a57064de32e@mail.gmail.com> Message-ID: <191c3a030906070732n677cb84k37941d026c8a401b@mail.gmail.com> Could you describe "latency"? not everyone uses it the same way. Maybe describe your exact problem. On Sat, Jun 6, 2009 at 6:46 PM, Nandy Dagondon wrote: > i'm using version build 13245M on an Intel D945GCLF2 Atom Dual-core mobo w/ > 2GB ram. > -nandy > > On Sat, Jun 6, 2009 at 11:02 AM, Brian West wrote: > >> You shouldn't be having problems... what version are you using? >> /b >> >> On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: >> >> there 10 client seats so at max. 10 simultaneous calls. however, the >> number of clients may be increased. >> -nandy >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/c09f5619/attachment-0002.html From anthony.minessale at gmail.com Sun Jun 7 07:36:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 7 Jun 2009 09:36:15 -0500 Subject: [Freeswitch-users] Problems subscribing to outbound socket events In-Reply-To: References: Message-ID: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> once you send "myevents" you lock on to only that channel's events despite any other *events* command. you may want to use the filter feature on the channel's uuid instead. On Sat, Jun 6, 2009 at 12:39 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I?ve put some c++ test code together to let the outbound socket control > the call, all works as expected, apart from the event subscription > > > > Sending myevents\n\n gives the channel events > > > > However sending event text all\n\n doesn?t give me any events apart from the channel events. > > > > > > Anyone care to suggest what I might be doing wrong? > > > > Regards, > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/23f4d011/attachment-0002.html From anthony.minessale at gmail.com Sun Jun 7 07:37:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 7 Jun 2009 09:37:40 -0500 Subject: [Freeswitch-users] How to reject a call without answering In-Reply-To: <20090606042846.72320@gmx.net> References: <20090606033426.72340@gmx.net> <5DF7AD0B-FC60-45CE-BB35-ABB1475E8E7D@freeswitch.org> <20090606042846.72320@gmx.net> Message-ID: <191c3a030906070737p5c791316ub30845321bfdf7e9@mail.gmail.com> the 2nd one. On Fri, Jun 5, 2009 at 11:28 PM, Klaus Teller wrote: > It doesn't seem to work. I tried the following: > > api respond 9015430e-82cf-418c-bf4c-f3ac6e85caf2 503 > > > SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 > call-command: execute > execute-app-name: respond > execute-app-arg: 503 > > Is one of these what you meant? > > Klaus. > > -------- Original-Nachricht -------- > > Datum: Fri, 5 Jun 2009 22:45:29 -0500 > > Von: Brian West > > An: freeswitch-users at lists.freeswitch.org > > Betreff: Re: [Freeswitch-users] How to reject a call without answering > > > Try the respond app. > > > > /b > > > > On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: > > > > > Hi, > > > > > > Going through the socket api, how can i reject a call without having > > > to answer it first? > > > > > > I tried sending a hangup command with cause set either to NO_ANSWER > > > or NORMAL_CLEARING. In both cases, Freeswitch does create another > > > socket to deliver the very same call. > > > > > > More precisely, when a call comes in i send a connect command. Then > > > after some few seconds, i then send the following hangup command: > > > > > > SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f > > > call-command: hangup > > > hangup-cause: NO_ANSWER > > > > > > Thanks for any feedback. > > > > > > Klaus. > > > > > > -- > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > -- > GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 > Euro/mtl.! > http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/9abb451c/attachment-0002.html From gerry at pstn2.net Sun Jun 7 09:31:19 2009 From: gerry at pstn2.net (Gerry Hull) Date: Sun, 7 Jun 2009 12:31:19 -0400 Subject: [Freeswitch-users] I need a favor... In-Reply-To: <4A2B7580.9020909@gmail.com> References: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> <4A2B7580.9020909@gmail.com> Message-ID: <98a86adf0906070931w71771cf3w83d12081f9eae568@mail.gmail.com> I have the installer project.. Can't get it to build. Can provIde ftp or will try ur project Thanks Gerry On 6/7/09, Meftah Tayeb wrote: > hello, > welcome, i'm able to build a Installer for your Freeswitch > please: > i have a very small Internet connection (128KBPS) and FS Setup syse is > 40MB or +... > i give you only the setup project and you compile it... ok? > thanks > Gerry Hull wrote: >> OK, thanks to help on the list have my very cool FreeSwitch app >> running... Gotta love FS once you get over the learning hump! >> >> So, I build FS and got everything running smoothly on my Wndows >> development box. Great. Then I went to deploy it on a production >> server. As I figured, no copy-and-run here. >> >> I tried building the setup project but it's just not happening for me! >> >> Can someone out there build me the Windows MSI for build 13496 or >> later and provide a link to it? I'm in a bind here to get this up >> and running. >> >> If I can pry a few bux out of the boss, I hope to be a ClueCon and >> describe to application we have built with FreeSwitch. >> >> Regards, >> >> Gerry >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > From brad.tuan at gmail.com Sun Jun 7 18:28:55 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Mon, 8 Jun 2009 09:28:55 +0800 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? Message-ID: As title, How to receive a call from another SIP proxy?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/8a57e2c9/attachment-0002.html From brian at freeswitch.org Sun Jun 7 18:40:12 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 7 Jun 2009 20:40:12 -0500 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? In-Reply-To: References: Message-ID: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> Thats a rather broad question... can you tell us if you have to register to said proxy? /b On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote: > As title, > > How to receive a call from another SIP proxy?? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090607/fa16c68c/attachment-0002.html From brad.tuan at gmail.com Sun Jun 7 19:00:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Mon, 8 Jun 2009 10:00:49 +0800 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? In-Reply-To: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> References: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> Message-ID: If myFS(203.64.xx.xx) want to receive a call from a SIP Proxy(163.28.xx.xx) In another word, User1(another SIP Proxy) want to call User2(FS), and User1 doesn't register to FS and User2 doesn't register to 163.28.xx.xx. How to set my FS?? 2009/6/8 Brian West > Thats a rather broad question... can you tell us if you have to register to > said proxy? > /b > > On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote: > > As title, > > How to receive a call from another SIP proxy?? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/502b86b0/attachment-0002.html From jmesquita at gmail.com Sun Jun 7 20:13:41 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 8 Jun 2009 00:13:41 -0300 Subject: [Freeswitch-users] How to receive a call from another SIP proxy?? In-Reply-To: References: <6816D452-CB66-4D01-9077-0277EF71EE75@freeswitch.org> Message-ID: <5a8712120906072013r16915308v2a67ae53253ba916@mail.gmail.com> That is what the Sofia External profile is for! On the default config, it is set to receive SIP messages on port 5080. You should give some study on the default configs.... Take a look at the files (read all the nice comments) and follow it with the wiki. That would be the road to fortune. Later, jmesquita On Sun, Jun 7, 2009 at 11:00 PM, Brad Tuan wrote: > If myFS(203.64.xx.xx) want to receive a call from a SIP Proxy(163.28.xx.xx) > > In another word, User1(another SIP Proxy) want to call User2(FS), > > and User1 doesn't register to FS and User2 doesn't register to > 163.28.xx.xx. > > How to set my FS?? > 2009/6/8 Brian West > >> Thats a rather broad question... can you tell us if you have to register >> to said proxy? >> /b >> >> On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote: >> >> As title, >> >> How to receive a call from another SIP proxy?? >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/bd0a5cd7/attachment-0002.html From dujinfang at gmail.com Sun Jun 7 23:29:44 2009 From: dujinfang at gmail.com (seven) Date: Mon, 8 Jun 2009 14:29:44 +0800 Subject: [Freeswitch-users] busy tone detect issue In-Reply-To: <7d0bfd8c0906052003w40a8fff4s8644cb9773e346b5@mail.gmail.com> References: <200906041628066407689@gmail.com> <14B93CB9-5750-441F-8456-645EA9F8E7D6@gmail.com> <87f2f3b90906042214m257978b1y85a7dedeb565c0e@mail.gmail.com> <8030F817-C66A-4AE9-B633-4E5F76568CF8@gmail.com> <7d0bfd8c0906052003w40a8fff4s8644cb9773e346b5@mail.gmail.com> Message-ID: <0E7D06F2-0964-4332-A5BF-440E723E47DF@gmail.com> On Jun 6, 2009, at 11:03 AM, Nandy Dagondon wrote: > dujinfang, > > hv u tried OSLEC? it's really reduced echo even on the cheapy X100P > card on *. oslec works w/ FS, too. Thanks, will try. :) > > > -nandy > > On Sat, Jun 6, 2009 at 10:11 AM, dujinfang > wrote: > > On Jun 5, 2009, at 1:14 PM, Michael Collins wrote: >> >> >> On Thu, Jun 4, 2009 at 6:59 PM, seven wrote: >> I'm using openzap analog with tone_detect, it works(conference not >> tested). however, according to the asterisk book, Kewlstart can >> detect the busy tone and disconnect the circuit. does anyone knows >> how to configure kewlstart with freeswitch/openzap? guess we don't >> need tone_detect then. >> >> Dujinfang, >> >> Your telco must support "kewlstart" signaling for this to be >> effective. The telco probably calls it something different, like >> "disconnect supervision" or "drop in loop current" or "battery >> reversal" or something like that. In any case, if the signaling is >> supported then you need to set up your zaptel.conf with the >> appropriate signaling type, which is either fxoks or fxsks. (I can >> never remember because zaptel does it backwards where if you have >> an FXO port then it uses FXS signaling but if you have an FXS port >> it uses FXO signaling. Stupidity, to be sure, so be aware of it.) >> > > 1) Don't know why but the similar zaptel.conf works on asterisk. I > guess tone_detect in FS is equivalent to busydetect=yes in > Asterisk(zapata.conf) . > > > zaptel.conf > # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) > fxsks=1 > fxsks=2 > fxsks=3 > fxsks=4 > > # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" > fxoks=5 > fxoks=6 > fxoks=7 > fxoks=8 > > # Global data > > loadzone = us > defaultzone = us > > > zapata.conf > > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=no > ;echotraining=800 > rxgain=0.0 > txgain=0.0 > group=0 > callgroup=1 > pickupgroup=1 > immediate=no > busydetect=yes > > I agree the FXO and FXS signaling is weird, why not they just match > the care name and reverse that internally? > > 2) Another essue is if I dial out from a FXO port from a local > extension(sip and zap), I can hear much echo on FreeSWITCH but not > much echo on asterisk. the zt.conf as below and I tried to change > the echo_cancel_level to 32 or 128 got no much difference. Is there > any equivalent configuration in FS like echocanccelwhenbridged=no in > asterisk? can I set busydetect and echocancelwhenbridged and other > options like this ? > > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > echo_cancel_level => 64 > > >> Find the sample zaptel.conf that comes with the zaptel package and >> search it for fxsks or fxoks and you'll see some notes on how to >> set it up for your analog trunks. >> >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/20ccf284/attachment-0002.html From rdenert at tng.de Sun Jun 7 23:38:26 2009 From: rdenert at tng.de (Rudolf Denert) Date: Mon, 8 Jun 2009 08:38:26 +0200 (CEST) Subject: [Freeswitch-users] Insufficient RTP stream In-Reply-To: Message-ID: <26244917.324011244443106580.JavaMail.root@zimbra.tng.de> @Brian West Hello, could I send you the traces from the wireshark directly to your e-mail address? The reason are the IP-addresses. I don't want to publish them to the whole world. Of course, you could send your answer to my question to freeswitch-users at lists.freeswitch.org. Best regards ----- Urspr?ngliche Mail ----- Von: "Brian West" An: freeswitch-users at lists.freeswitch.org Gesendet: Donnerstag, 4. Juni 2009 16:13:29 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Insufficient RTP stream Sorry but this type of trace is impossible to read. I want raw pcap if possible. /b On Jun 4, 2009, at 2:52 AM, Rudolf Denert wrote: Ok, here is the SIP trace. If you need more, just tell me and I will send them. The RTP trace you already have, haven't you? Greetz Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From rdenert at tng.de Sun Jun 7 23:42:19 2009 From: rdenert at tng.de (Rudolf Denert) Date: Mon, 8 Jun 2009 08:42:19 +0200 (CEST) Subject: [Freeswitch-users] DTMF Problems Message-ID: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> Hello! Is there a possibility to "detect" or "scan" which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? Greetz From durk.debeer at isp.solcon.nl Mon Jun 8 00:37:16 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Mon, 08 Jun 2009 09:37:16 +0200 Subject: [Freeswitch-users] Problem with attendant transfer Message-ID: Hello all, I have observed an issue on using Freeswitch and some SIP-phones. Ok the problem is this. Some phones, when attempting an attendant transfer, put the recipient of the transfer on hold. This results in Freeswitch starting to stream MOH music to the phone put on hold, if implemented. When now the original phone is pasing the transfer, Freeswitch is not going to process this transfer because the recipient end of it is on hold. It is however terminating the connections it has with the phone initiating the transfer. This means that the recipient of the transfer is never coming of hold again until it terminates the call. Is there a way to detect this behaviour, so I can get the recipient of hold before Freeswitch is processing the transfer?. Kind regards Durk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/d42f6604/attachment.html From brian at freeswitch.org Mon Jun 8 07:26:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 09:26:14 -0500 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> References: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> Message-ID: <93F60015-4D72-474C-BF5D-B5EBA7574D4C@freeswitch.org> This makes no sense.... Can you try to explain it more? I do attended transfers with sip phones every day without a single problem. Maybe i'm missing what you're talking about. /b On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote: > Hello all, > I have observed an issue on using Freeswitch and some SIP-phones. Ok > the problem is this. Some phones, when attempting an attendant > transfer, put the recipient of the transfer on hold. This results in > Freeswitch starting to stream MOH music to the phone put on hold, if > implemented. When now the original phone is pasing the transfer, > Freeswitch is not going to process this transfer because the > recipient end of it is on hold. It is however terminating the > connections it has with the phone initiating the transfer. This > means that the recipient of the transfer is never coming of hold > again until it terminates the call. > Is there a way to detect this behaviour, so I can get the recipient > of hold before Freeswitch is processing the transfer?. > Kind regards > Durk Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/6a0de4e6/attachment-0002.html From klaus.teller at gmx.net Mon Jun 8 07:27:37 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Mon, 08 Jun 2009 16:27:37 +0200 Subject: [Freeswitch-users] Taking long at startup Message-ID: <20090608142737.272910@gmx.net> Hi, Freeswitch is taking quiet some time to start. Is is normal these days? it didn't used to be the case few months ago. Is there anything i can turn off to start faster? Thanks, Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From intralanman at freeswitch.org Mon Jun 8 11:34:56 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 08 Jun 2009 14:34:56 -0400 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> References: <4a2cc0fa.8653f10a.1a29.ffffad09SMTPIN_ADDED@mx.google.com> Message-ID: <4A2D59D0.1020407@freeswitch.org> Durk de Beer wrote: > > Hello all, > > I have observed an issue on using Freeswitch and some SIP-phones. Ok > the problem is this. Some phones > which phones? -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/cb1ff4e2/attachment-0002.html From brian at freeswitch.org Mon Jun 8 07:38:06 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 09:38:06 -0500 Subject: [Freeswitch-users] Taking long at startup In-Reply-To: <20090608142737.272910@gmx.net> References: <20090608142737.272910@gmx.net> Message-ID: <7E9EE8DC-9C8F-4D4A-80F2-36AFDCAD83C5@freeswitch.org> Update.. if it takes longer than 8 seconds start with -nonat /b On Jun 8, 2009, at 9:27 AM, Klaus Teller wrote: > Hi, > > Freeswitch is taking quiet some time to start. Is is normal these > days? it didn't used to be the case few months ago. Is there > anything i can turn off to start faster? > > Thanks, > Klaus. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/a4b06cc3/attachment-0002.html From peter.olsson at visionutveckling.se Mon Jun 8 07:42:35 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 16:42:35 +0200 Subject: [Freeswitch-users] Taking long at startup In-Reply-To: <20090608142737.272910@gmx.net> References: <20090608142737.272910@gmx.net> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB88E4@cooper> Klaus, This is probably caused by the new nat features introduced in FreeSWITCH. You can start FS with -nonat to skip this detection. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Klaus Teller Skickat: den 8 juni 2009 16:28 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Taking long at startup Hi, Freeswitch is taking quiet some time to start. Is is normal these days? it didn't used to be the case few months ago. Is there anything i can turn off to start faster? Thanks, Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d22ba32931975679412! From kristian.kielhofner at gmail.com Mon Jun 8 08:08:59 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 8 Jun 2009 11:08:59 -0400 Subject: [Freeswitch-users] DTMF Problems In-Reply-To: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> References: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> Message-ID: <2d9149cd0906080808k792dd3d6i5af9282e4dce02ab@mail.gmail.com> Rudolf, I believe there is a snippet in the sample XML dialplan to detect the lack of telephone-event in the SDP and activate inband detection. You could use that for inspiration. On Mon, Jun 8, 2009 at 2:42 AM, Rudolf Denert wrote: > Hello! > > Is there a possibility to "detect" or "scan" which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? > > Greetz > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Jun 8 08:12:35 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 10:12:35 -0500 Subject: [Freeswitch-users] DTMF Problems In-Reply-To: <2d9149cd0906080808k792dd3d6i5af9282e4dce02ab@mail.gmail.com> References: <14789954.324041244443339647.JavaMail.root@zimbra.tng.de> <2d9149cd0906080808k792dd3d6i5af9282e4dce02ab@mail.gmail.com> Message-ID: I wrote that to demonstrate that exact situation but you still can't tell if they are inband or info :P /b On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote: > Rudolf, > > I believe there is a snippet in the sample XML dialplan to detect > the lack of telephone-event in the SDP and activate inband detection. > You could use that for inspiration. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/aa9873b4/attachment-0002.html From rex.alex345 at yahoo.com Mon Jun 8 08:32:47 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 08:32:47 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> Message-ID: <1244475167805-3043665.post@n2.nabble.com> Hi Brian, I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 Addions made are, acl.conf.xml freeswitch.xml sip_profiles/external.xml under tag public.xml default.xml Still, Inbound is not hitting my FS console itself. Please assist where am I going wrong? Thanks, Rex Brian West wrote: > > > On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >> >> Hello, >> >> My public.xml configration is: >> >> >> >> >> >> > > $1 will not exist in this case because your regular expression doesn't > capture anything. So replace $1 with your target number or use > ^(123456)$ > > >> >> My default.xml configration is: >> >> >> >> >> >> >> >> > > Can you elaborate how you're registering with your provider? > > >> >> >> When I am trying to call 123456 from my mobile no. Not able to see any >> logging in FS console. Please assist where I am going wrong? Or do I >> require >> any extra modules to be installed? >> >> Thanks, >> Rex > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rex.alex345 at yahoo.com Mon Jun 8 08:35:50 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 08:35:50 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244475167805-3043665.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <0012C2D7-4DAD-4A39-9F80-AFF8FA5B7839@gmail.com> <1243959110713-3012928.post@n2.nabble.com> <1244475167805-3043665.post@n2.nabble.com> Message-ID: <1244475350818-3043700.post@n2.nabble.com> Hi Brian, Missed.... sip_profiles/external.xml under tag Thanks, Rex Rex_Alex wrote: > > Hi Brian, > > I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > > Addions made are, > > acl.conf.xml > > > > > > freeswitch.xml > > > > sip_profiles/external.xml > > under tag > > public.xml > > > > > > > > default.xml > > > > > > > > > > Still, Inbound is not hitting my FS console itself. Please assist where am > I going wrong? > > Thanks, > Rex > > > > > > > > Brian West wrote: >> >> >> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >> >>> >>> Hello, >>> >>> My public.xml configration is: >>> >>> >>> >>> >>> >>> >> >> $1 will not exist in this case because your regular expression doesn't >> capture anything. So replace $1 with your target number or use >> ^(123456)$ >> >> >>> >>> My default.xml configration is: >>> >>> >>> >>> >>> >>> >>> >>> >> >> Can you elaborate how you're registering with your provider? >> >> >>> >>> >>> When I am trying to call 123456 from my mobile no. Not able to see any >>> logging in FS console. Please assist where I am going wrong? Or do I >>> require >>> any extra modules to be installed? >>> >>> Thanks, >>> Rex >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043700.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 08:44:19 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 17:44:19 +0200 Subject: [Freeswitch-users] Inbound using FS Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> Have you configured the sip profile to use the acl list you have created (Inbound_Test)? /Peter ----- Ursprungligt meddelande ----- Fr?n: Rex_Alex Skickat: den 8 juni 2009 17:40 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Inbound using FS Hi Brian, I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 Addions made are, acl.conf.xml freeswitch.xml sip_profiles/external.xml under tag public.xml default.xml Still, Inbound is not hitting my FS console itself. Please assist where am I going wrong? Thanks, Rex Brian West wrote: > > > On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >> >> Hello, >> >> My public.xml configration is: >> >> >> >> >> >> > > $1 will not exist in this case because your regular expression doesn't > capture anything. So replace $1 with your target number or use > ^(123456)$ > > >> >> My default.xml configration is: >> >> >> >> >> >> >> >> > > Can you elaborate how you're registering with your provider? > > >> >> >> When I am trying to call 123456 from my mobile no. Not able to see any >> logging in FS console. Please assist where I am going wrong? Or do I >> require >> any extra modules to be installed? >> >> Thanks, >> Rex > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d30e032931222027793! From rex.alex345 at yahoo.com Mon Jun 8 08:53:30 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 08:53:30 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> Message-ID: <1244476410874-3043804.post@n2.nabble.com> Hello Peter, Yes, I have added under tag in sip_profiles/external.xml Thanks, Rex Peter Olsson wrote: > > Have you configured the sip profile to use the acl list you have created > (Inbound_Test)? > > /Peter > > > ----- Ursprungligt meddelande ----- > Fr?n: Rex_Alex > Skickat: den 8 juni 2009 17:40 > Till: freeswitch-users at lists.freeswitch.org > > ?mne: Re: [Freeswitch-users] Inbound using FS > > > Hi Brian, > > I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > > Addions made are, > > acl.conf.xml > > > > > > freeswitch.xml > > > > sip_profiles/external.xml > > under tag > > public.xml > > > > > > > > default.xml > > > > > > data="sofia/internal/1007%1.1.1.1"/> > > > > Still, Inbound is not hitting my FS console itself. Please assist where am > I > going wrong? > > Thanks, > Rex > > > > > > > > Brian West wrote: >> >> >> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >> >>> >>> Hello, >>> >>> My public.xml configration is: >>> >>> >>> >>> >>> >>> >> >> $1 will not exist in this case because your regular expression doesn't >> capture anything. So replace $1 with your target number or use >> ^(123456)$ >> >> >>> >>> My default.xml configration is: >>> >>> >>> >>> >>> >>> >>> >>> >> >> Can you elaborate how you're registering with your provider? >> >> >>> >>> >>> When I am trying to call 123456 from my mobile no. Not able to see any >>> logging in FS console. Please assist where I am going wrong? Or do I >>> require >>> any extra modules to be installed? >>> >>> Thanks, >>> Rex >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d30e032931222027793! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 09:18:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 18:18:28 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244476410874-3043804.post@n2.nabble.com> Message-ID: I don't see what you've added. But I guess it's something like . Are you sure you're dialing into the external profile? It's on port 5080 by default, and the internal is on 5060. /Peter On 09-06-08 17.53, "Rex_Alex" wrote: Hello Peter, Yes, I have added under tag in sip_profiles/external.xml Thanks, Rex Peter Olsson wrote: > > Have you configured the sip profile to use the acl list you have created > (Inbound_Test)? > > /Peter > > > ----- Ursprungligt meddelande ----- > Fr?n: Rex_Alex > Skickat: den 8 juni 2009 17:40 > Till: freeswitch-users at lists.freeswitch.org > > ?mne: Re: [Freeswitch-users] Inbound using FS > > > Hi Brian, > > I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > > Addions made are, > > acl.conf.xml > > > > > > freeswitch.xml > > > > sip_profiles/external.xml > > under tag > > public.xml > > > > > > > > default.xml > > > > > > data="sofia/internal/1007%1.1.1.1"/> > > > > Still, Inbound is not hitting my FS console itself. Please assist where am > I > going wrong? > > Thanks, > Rex > > > > > > > > Brian West wrote: >> >> >> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >> >>> >>> Hello, >>> >>> My public.xml configration is: >>> >>> >>> >>> >>> >>> >> >> $1 will not exist in this case because your regular expression doesn't >> capture anything. So replace $1 with your target number or use >> ^(123456)$ >> >> >>> >>> My default.xml configration is: >>> >>> >>> >>> >>> >>> >>> >>> >> >> Can you elaborate how you're registering with your provider? >> >> >>> >>> >>> When I am trying to call 123456 from my mobile no. Not able to see any >>> logging in FS console. Please assist where I am going wrong? Or do I >>> require >>> any extra modules to be installed? >>> >>> Thanks, >>> Rex >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d354232931305258464! From rex.alex345 at yahoo.com Mon Jun 8 09:34:36 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 09:34:36 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> Message-ID: <1244478876050-3044052.post@n2.nabble.com> Hello, I am not sure about the profile which I am calling into. My scenario is this, I am trying to reach extn 1007 registered in FS server from my mobile through an inbound PRI connected to the audiocode with DID 123456. Thanks, Rex Peter Olsson wrote: > > I don't see what you've added. But I guess it's something like . > > Are you sure you're dialing into the external profile? It's on port 5080 > by default, and the internal is on 5060. > > /Peter > > > On 09-06-08 17.53, "Rex_Alex" wrote: > > > > Hello Peter, > > Yes, I have added > > > > under tag in sip_profiles/external.xml > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> Have you configured the sip profile to use the acl list you have created >> (Inbound_Test)? >> >> /Peter >> >> >> ----- Ursprungligt meddelande ----- >> Fr?n: Rex_Alex >> Skickat: den 8 juni 2009 17:40 >> Till: freeswitch-users at lists.freeswitch.org >> >> ?mne: Re: [Freeswitch-users] Inbound using FS >> >> >> Hi Brian, >> >> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >> >> Addions made are, >> >> acl.conf.xml >> >> >> >> >> >> freeswitch.xml >> >> >> >> sip_profiles/external.xml >> >> under tag >> >> public.xml >> >> >> >> >> >> >> >> default.xml >> >> >> >> >> >> > data="sofia/internal/1007%1.1.1.1"/> >> >> >> >> Still, Inbound is not hitting my FS console itself. Please assist where >> am >> I >> going wrong? >> >> Thanks, >> Rex >> >> >> >> >> >> >> >> Brian West wrote: >>> >>> >>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>> >>>> >>>> Hello, >>>> >>>> My public.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> $1 will not exist in this case because your regular expression doesn't >>> capture anything. So replace $1 with your target number or use >>> ^(123456)$ >>> >>> >>>> >>>> My default.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> Can you elaborate how you're registering with your provider? >>> >>> >>>> >>>> >>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>> logging in FS console. Please assist where I am going wrong? Or do I >>>> require >>>> any extra modules to be installed? >>>> >>>> Thanks, >>>> Rex >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d354232931305258464! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 09:50:57 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 18:50:57 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244478876050-3044052.post@n2.nabble.com> Message-ID: The PRI/SIP-box probably talks to the internal profile (I guess that they are on the same LAN). Try to add the inbound-acl to the internal profile as well. Also restart FS completely, just to be 100% sure that config is reloaded. //Peter On 09-06-08 18.34, "Rex_Alex" wrote: Hello, I am not sure about the profile which I am calling into. My scenario is this, I am trying to reach extn 1007 registered in FS server from my mobile through an inbound PRI connected to the audiocode with DID 123456. Thanks, Rex Peter Olsson wrote: > > I don't see what you've added. But I guess it's something like . > > Are you sure you're dialing into the external profile? It's on port 5080 > by default, and the internal is on 5060. > > /Peter > > > On 09-06-08 17.53, "Rex_Alex" wrote: > > > > Hello Peter, > > Yes, I have added > > > > under tag in sip_profiles/external.xml > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> Have you configured the sip profile to use the acl list you have created >> (Inbound_Test)? >> >> /Peter >> >> >> ----- Ursprungligt meddelande ----- >> Fr?n: Rex_Alex >> Skickat: den 8 juni 2009 17:40 >> Till: freeswitch-users at lists.freeswitch.org >> >> ?mne: Re: [Freeswitch-users] Inbound using FS >> >> >> Hi Brian, >> >> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >> >> Addions made are, >> >> acl.conf.xml >> >> >> >> >> >> freeswitch.xml >> >> >> >> sip_profiles/external.xml >> >> under tag >> >> public.xml >> >> >> >> >> >> >> >> default.xml >> >> >> >> >> >> > data="sofia/internal/1007%1.1.1.1"/> >> >> >> >> Still, Inbound is not hitting my FS console itself. Please assist where >> am >> I >> going wrong? >> >> Thanks, >> Rex >> >> >> >> >> >> >> >> Brian West wrote: >>> >>> >>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>> >>>> >>>> Hello, >>>> >>>> My public.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> $1 will not exist in this case because your regular expression doesn't >>> capture anything. So replace $1 with your target number or use >>> ^(123456)$ >>> >>> >>>> >>>> My default.xml configration is: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> Can you elaborate how you're registering with your provider? >>> >>> >>>> >>>> >>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>> logging in FS console. Please assist where I am going wrong? Or do I >>>> require >>>> any extra modules to be installed? >>>> >>>> Thanks, >>>> Rex >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d3f8432938250412368! From rex.alex345 at yahoo.com Mon Jun 8 10:04:36 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Mon, 8 Jun 2009 10:04:36 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> Message-ID: <1244480676920-3044219.post@n2.nabble.com> Hello, Yes you are right. they are the on the same LAN. Inboun-acl added in internal profile as well and restarted the FS completely. But no luck.. Please help us to resolve the same.. Thanks, Rex Peter Olsson wrote: > > The PRI/SIP-box probably talks to the internal profile (I guess that they > are on the same LAN). Try to add the inbound-acl to the internal profile > as well. Also restart FS completely, just to be 100% sure that config is > reloaded. > > //Peter > > > On 09-06-08 18.34, "Rex_Alex" wrote: > > > > Hello, > > I am not sure about the profile which I am calling into. > > My scenario is this, I am trying to reach extn 1007 registered in FS > server > from my mobile through an inbound PRI connected to the audiocode with DID > 123456. > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> I don't see what you've added. But I guess it's something like . >> >> Are you sure you're dialing into the external profile? It's on port 5080 >> by default, and the internal is on 5060. >> >> /Peter >> >> >> On 09-06-08 17.53, "Rex_Alex" wrote: >> >> >> >> Hello Peter, >> >> Yes, I have added >> >> >> >> under tag in sip_profiles/external.xml >> >> Thanks, >> Rex >> >> >> >> Peter Olsson wrote: >>> >>> Have you configured the sip profile to use the acl list you have created >>> (Inbound_Test)? >>> >>> /Peter >>> >>> >>> ----- Ursprungligt meddelande ----- >>> Fr?n: Rex_Alex >>> Skickat: den 8 juni 2009 17:40 >>> Till: freeswitch-users at lists.freeswitch.org >>> >>> ?mne: Re: [Freeswitch-users] Inbound using FS >>> >>> >>> Hi Brian, >>> >>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >>> >>> Addions made are, >>> >>> acl.conf.xml >>> >>> >>> >>> >>> >>> freeswitch.xml >>> >>> >>> >>> sip_profiles/external.xml >>> >>> under tag >>> >>> public.xml >>> >>> >>> >>> >>> >>> >>> >>> default.xml >>> >>> >>> >>> >>> >>> >> data="sofia/internal/1007%1.1.1.1"/> >>> >>> >>> >>> Still, Inbound is not hitting my FS console itself. Please assist where >>> am >>> I >>> going wrong? >>> >>> Thanks, >>> Rex >>> >>> >>> >>> >>> >>> >>> >>> Brian West wrote: >>>> >>>> >>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>>> >>>>> >>>>> Hello, >>>>> >>>>> My public.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> $1 will not exist in this case because your regular expression doesn't >>>> capture anything. So replace $1 with your target number or use >>>> ^(123456)$ >>>> >>>> >>>>> >>>>> My default.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> Can you elaborate how you're registering with your provider? >>>> >>>> >>>>> >>>>> >>>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>>> logging in FS console. Please assist where I am going wrong? Or do I >>>>> require >>>>> any extra modules to be installed? >>>>> >>>>> Thanks, >>>>> Rex >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d3f8432938250412368! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jun 8 10:22:43 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 8 Jun 2009 19:22:43 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244480676920-3044219.post@n2.nabble.com> Message-ID: Are you able to see anything at all in the console/log? I'm starting to doubt that the call even gets into the FS box... :) Try enabling more logs (console loglevel debug), and try again. On 09-06-08 19.04, "Rex_Alex" wrote: Hello, Yes you are right. they are the on the same LAN. Inboun-acl added in internal profile as well and restarted the FS completely. But no luck.. Please help us to resolve the same.. Thanks, Rex Peter Olsson wrote: > > The PRI/SIP-box probably talks to the internal profile (I guess that they > are on the same LAN). Try to add the inbound-acl to the internal profile > as well. Also restart FS completely, just to be 100% sure that config is > reloaded. > > //Peter > > > On 09-06-08 18.34, "Rex_Alex" wrote: > > > > Hello, > > I am not sure about the profile which I am calling into. > > My scenario is this, I am trying to reach extn 1007 registered in FS > server > from my mobile through an inbound PRI connected to the audiocode with DID > 123456. > > Thanks, > Rex > > > > Peter Olsson wrote: >> >> I don't see what you've added. But I guess it's something like . >> >> Are you sure you're dialing into the external profile? It's on port 5080 >> by default, and the internal is on 5060. >> >> /Peter >> >> >> On 09-06-08 17.53, "Rex_Alex" wrote: >> >> >> >> Hello Peter, >> >> Yes, I have added >> >> >> >> under tag in sip_profiles/external.xml >> >> Thanks, >> Rex >> >> >> >> Peter Olsson wrote: >>> >>> Have you configured the sip profile to use the acl list you have created >>> (Inbound_Test)? >>> >>> /Peter >>> >>> >>> ----- Ursprungligt meddelande ----- >>> Fr?n: Rex_Alex >>> Skickat: den 8 juni 2009 17:40 >>> Till: freeswitch-users at lists.freeswitch.org >>> >>> ?mne: Re: [Freeswitch-users] Inbound using FS >>> >>> >>> Hi Brian, >>> >>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >>> >>> Addions made are, >>> >>> acl.conf.xml >>> >>> >>> >>> >>> >>> freeswitch.xml >>> >>> >>> >>> sip_profiles/external.xml >>> >>> under tag >>> >>> public.xml >>> >>> >>> >>> >>> >>> >>> >>> default.xml >>> >>> >>> >>> >>> >>> >> data="sofia/internal/1007%1.1.1.1"/> >>> >>> >>> >>> Still, Inbound is not hitting my FS console itself. Please assist where >>> am >>> I >>> going wrong? >>> >>> Thanks, >>> Rex >>> >>> >>> >>> >>> >>> >>> >>> Brian West wrote: >>>> >>>> >>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>>> >>>>> >>>>> Hello, >>>>> >>>>> My public.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> $1 will not exist in this case because your regular expression doesn't >>>> capture anything. So replace $1 with your target number or use >>>> ^(123456)$ >>>> >>>> >>>>> >>>>> My default.xml configration is: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> Can you elaborate how you're registering with your provider? >>>> >>>> >>>>> >>>>> >>>>> When I am trying to call 123456 from my mobile no. Not able to see any >>>>> logging in FS console. Please assist where I am going wrong? Or do I >>>>> require >>>>> any extra modules to be installed? >>>>> >>>>> Thanks, >>>>> Rex >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2d45dc32931463593608! From timb0311 at hotmail.com Mon Jun 8 11:31:44 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 8 Jun 2009 14:31:44 -0400 Subject: [Freeswitch-users] Routing 911 calls In-Reply-To: References: Message-ID: If I have 1 inbound DID and "client software" that calls this DID from multiple locations (states, city, etc), how would I go about placing and routing a 911 / emergency calls? Keep in mind each client does have their own existing phone line (DID) with 911/e911 attached to that DID. Could I route by setting outbound caller id and number to the incoming, and then place the outbound 911 call? Tim _________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/ade2bb75/attachment-0002.html From brian at freeswitch.org Mon Jun 8 11:43:37 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 13:43:37 -0500 Subject: [Freeswitch-users] Routing 911 calls In-Reply-To: References: Message-ID: Not sure that'll meet the legal requirements. /b On Jun 8, 2009, at 1:31 PM, Tim B wrote: > Could I route by setting outbound caller id and number to the > incoming, and then place the outbound 911 call? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/ae224788/attachment-0002.html From msc at freeswitch.org Mon Jun 8 12:03:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 8 Jun 2009 12:03:55 -0700 Subject: [Freeswitch-users] Routing 911 calls In-Reply-To: References: Message-ID: <87f2f3b90906081203w595f4f56k19579b7459e72ab3@mail.gmail.com> On Mon, Jun 8, 2009 at 11:31 AM, Tim B wrote: > If I have 1 inbound DID and "client software" that calls this DID from > multiple locations (states, city, etc), how would I go about placing and > routing a 911 / emergency calls? Keep in mind each client does have their > own existing phone line (DID) with 911/e911 attached to that DID. > > Could I route by setting outbound caller id and number to the incoming, and > then place the outbound 911 call? > > You can try it, and then make a test 911 call and tell the operator that you are testing a new phone system install. The 911 op can then verify the information he/she sees on the screen with where you are. If you're in New York when you call but the 911 op sees "Jerkwater, Alabama" then you know it doesn't work... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/599843e3/attachment-0002.html From brian at freeswitch.org Mon Jun 8 12:08:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 14:08:40 -0500 Subject: [Freeswitch-users] Sounds for 1.0.4 release and zRTP sound files. Message-ID: I have a deal from GM Voices for this order at a major discount but its still $650 USD, If you wish to donate to this order please let me know... brian at freeswitch.org is my paypal. Thanks, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/5049a2d2/attachment-0002.html From c_cav_01 at yahoo.com Mon Jun 8 13:18:49 2009 From: c_cav_01 at yahoo.com (Chris) Date: Mon, 8 Jun 2009 13:18:49 -0700 (PDT) Subject: [Freeswitch-users] Routing 911 calls Message-ID: <796626.25975.qm@web55103.mail.re4.yahoo.com> Looking for a reasonable DID/e911 provider.? Any suggestions. p.s.? Sorry to hijack the thread.? --- On Mon, 6/8/09, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] Routing 911 calls To: freeswitch-users at lists.freeswitch.org Date: Monday, June 8, 2009, 1:03 PM On Mon, Jun 8, 2009 at 11:31 AM, Tim B wrote: If I have 1 inbound DID and "client software"?that calls this DID from multiple locations (states, city, etc), how would I go about placing and routing?a 911 / emergency?calls?? Keep in mind each client does have their own existing phone line (DID)?with 911/e911 attached to that DID. ? Could I route by setting outbound caller id and number to the incoming, and then place the outbound 911 call? ? You can try it, and then make a test 911 call and tell the operator that you are testing a new phone system install. The 911 op can then verify the information he/she sees on the screen with where you are. If you're in New York when you call but the 911 op sees "Jerkwater, Alabama" then you know it doesn't work... -MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/5a82a92a/attachment-0002.html From msc at freeswitch.org Mon Jun 8 14:31:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 8 Jun 2009 14:31:23 -0700 Subject: [Freeswitch-users] Problems subscribing to outbound socket events In-Reply-To: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> References: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> Message-ID: <87f2f3b90906081431n38500f1fi35ee7e5d314733d7@mail.gmail.com> On Sun, Jun 7, 2009 at 7:36 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > once you send "myevents" you lock on to only that channel's events despite > any other *events* command. > you may want to use the filter feature on the channel's uuid instead. > > FYI, I've added this information to the event socket wiki page. There wasn't an entry on the filter command so I added one with some simple examples. Hope this helps people in the future. http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/8413e8e2/attachment-0002.html From jmesquita at gmail.com Mon Jun 8 15:33:41 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 8 Jun 2009 19:33:41 -0300 Subject: [Freeswitch-users] Problems subscribing to outbound socket events In-Reply-To: <87f2f3b90906081431n38500f1fi35ee7e5d314733d7@mail.gmail.com> References: <191c3a030906070736u513bcfeduf7d136a075563fd1@mail.gmail.com> <87f2f3b90906081431n38500f1fi35ee7e5d314733d7@mail.gmail.com> Message-ID: <5a8712120906081533p198d3c2bq2a758f774a0f986e@mail.gmail.com> MC is tha man... jmesquita On Mon, Jun 8, 2009 at 6:31 PM, Michael Collins wrote: > > > On Sun, Jun 7, 2009 at 7:36 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> once you send "myevents" you lock on to only that channel's events despite >> any other *events* command. >> you may want to use the filter feature on the channel's uuid instead. >> >> > FYI, I've added this information to the event socket wiki page. There > wasn't an entry on the filter command so I added one with some simple > examples. Hope this helps people in the future. > > http://wiki.freeswitch.org/wiki/Mod_event_socket#filter > > -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/a55f50d3/attachment-0002.html From john at feith.com Mon Jun 8 17:40:22 2009 From: john at feith.com (John Wehle) Date: Mon, 8 Jun 2009 20:40:22 -0400 (EDT) Subject: [Freeswitch-users] Caller id when doing transfers Message-ID: <200906090040.n590eMHr004215@jwlab.FEITH.COM> Consider the following sequence: 1) Outside caller (OC) calls Ext 1001 Caller id shows OC 2) Ext 1001 transfers the call to Ext 1002 In some cases we want the caller id to shows OC, in other cases we want the caller id to show Ext 1001. It appears from some limited testing that the original caller id is always shown when the call is transfered. Is there some way to have the person making the transfer show up as the caller id? Our application is I want to setup an extension (*5) which automatically places calls into a fifo corresponding to the Extension number of the person transferring the call. This will provide park capability similar to that of our old System 25 PBX. I'll then setup an extension (*8) which picks up the call at the front of the fifo "fifo". -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From jim at evolutiontel.net Mon Jun 8 17:44:00 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 9 Jun 2009 10:44:00 +1000 Subject: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. In-Reply-To: <51DECC6F-E822-4459-BF47-970576041825@freeswitch.org> References: <51DECC6F-E822-4459-BF47-970576041825@freeswitch.org> Message-ID: Gents, Thanks for your input on this.....much appreciated. Changing the codec from G729a to G729 before the call hits FS did the trick. Calls are now going through correctly. Cheers, Jim On Sat, Jun 6, 2009 at 12:32 PM, Brian West wrote: > G729a is 100% INVALID in the sdp on codec 18. > /b > On Jun 5, 2009, at 9:25 PM, Jim Burke wrote: > > Noticed there is a list of mime types associated with FS and G729a was not > listed, does this have anything to do with the root cause? > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net From larclap at yahoo.com Mon Jun 8 19:24:46 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 8 Jun 2009 19:24:46 -0700 Subject: [Freeswitch-users] Can't hear outbound calls Message-ID: <005001c9e8a9$76c5c780$64515680$@com> I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/869885cf/attachment-0002.html From brian at freeswitch.org Mon Jun 8 19:38:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 8 Jun 2009 21:38:39 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <005001c9e8a9$76c5c780$64515680$@com> References: <005001c9e8a9$76c5c780$64515680$@com> Message-ID: <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> pastebin your profile config and the output of global_getvar /b On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: > I had a working FS installation which I messed up by doing a fresh > install. I tried to integrate all my custom changes, but I?m sure I > screwed something up. > > The symptom is on an outbound call, sometimes I can hear ringing, > other times I cannot. Finally I can see FS connects via a softphone, > but I hear only silence. The other side of the conversation hears > static. > > I pasted a siptrace of the external profile. The Contact, Via and > SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 > > FS exists on a LAN behind a NAT firewall along with all its clients. > There is a SwitchVox system which predates the FS. I had to use an > external sip = 5090 for FS. Also I think I had to use a different > WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) > than the one FS came up with (xxx.xxx.xxx.82), but I can?t figure > out where I set this address. > > I would appreciate any help. Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/1e9006e9/attachment-0002.html From larclap at yahoo.com Mon Jun 8 19:57:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 8 Jun 2009 19:57:45 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> Message-ID: <006101c9e8ae$11f03c00$35d0b400$@com> http://pastebin.freeswitch.org/9319 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 08, 2009 7:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls pastebin your profile config and the output of global_getvar /b On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090608/9e2f9a3a/attachment-0002.html From jason at jasonjgw.net Mon Jun 8 20:30:51 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 9 Jun 2009 13:30:51 +1000 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <005001c9e8a9$76c5c780$64515680$@com> References: <005001c9e8a9$76c5c780$64515680$@com> Message-ID: <20090609033051.GA28848@jdc.jasonjgw.net> Lars Zeb wrote: > I had a working FS installation which I messed up by doing a fresh install. > I tried to integrate all my custom changes, but I'm sure I screwed something > up. Git is an excellent tool for keeping track of FreeSWITCH configuration changes. The history of my configuration is maintained in a git repository under /opt/freeswitch/conf - git simply creates a .git subdirectory to store all of the revisions as they are committed. Git revert and git stash have been very useful at times, not to mention git reset --hard. Since Git is used for Linux kernel development, it should be available from most recent Linux distributions, and it can probably be compiled for other Unix-like environments as well. From saigop at gmail.com Mon Jun 8 21:47:21 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Tue, 9 Jun 2009 10:17:21 +0530 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1244480676920-3044219.post@n2.nabble.com> Message-ID: <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> Hi Rex, You need to allow your acl in internal.xml like the one, Change the internal-network according to your configuration you allowed in acl.conf.xml. I have tested with audiocode with PRI line its working fine. On Mon, Jun 8, 2009 at 10:52 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Are you able to see anything at all in the console/log? > > I'm starting to doubt that the call even gets into the FS box... :) > > Try enabling more logs (console loglevel debug), and try again. > > > On 09-06-08 19.04, "Rex_Alex" wrote: > > > > > > Hello, > > Yes you are right. they are the on the same LAN. Inboun-acl added > in internal profile as well and restarted the FS completely. But no luck.. > Please help us to resolve the same.. > > Thanks, > Rex > > > Peter Olsson wrote: > > > > The PRI/SIP-box probably talks to the internal profile (I guess that they > > are on the same LAN). Try to add the inbound-acl to the internal profile > > as well. Also restart FS completely, just to be 100% sure that config is > > reloaded. > > > > //Peter > > > > > > On 09-06-08 18.34, "Rex_Alex" wrote: > > > > > > > > Hello, > > > > I am not sure about the profile which I am calling into. > > > > My scenario is this, I am trying to reach extn 1007 registered in FS > > server > > from my mobile through an inbound PRI connected to the audiocode with DID > > 123456. > > > > Thanks, > > Rex > > > > > > > > Peter Olsson wrote: > >> > >> I don't see what you've added. But I guess it's something like . > >> > >> Are you sure you're dialing into the external profile? It's on port 5080 > >> by default, and the internal is on 5060. > >> > >> /Peter > >> > >> > >> On 09-06-08 17.53, "Rex_Alex" wrote: > >> > >> > >> > >> Hello Peter, > >> > >> Yes, I have added > >> > >> > >> > >> under tag in sip_profiles/external.xml > >> > >> Thanks, > >> Rex > >> > >> > >> > >> Peter Olsson wrote: > >>> > >>> Have you configured the sip profile to use the acl list you have > created > >>> (Inbound_Test)? > >>> > >>> /Peter > >>> > >>> > >>> ----- Ursprungligt meddelande ----- > >>> Fr?n: Rex_Alex > >>> Skickat: den 8 juni 2009 17:40 > >>> Till: freeswitch-users at lists.freeswitch.org > >>> > >>> ?mne: Re: [Freeswitch-users] Inbound using FS > >>> > >>> > >>> Hi Brian, > >>> > >>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 > >>> > >>> Addions made are, > >>> > >>> acl.conf.xml > >>> > >>> > >>> > >>> > >>> > >>> freeswitch.xml > >>> > >>> > >>> > >>> sip_profiles/external.xml > >>> > >>> under tag > >>> > >>> public.xml > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> default.xml > >>> > >>> > >>> > >>> > >>> > >>> >>> data="sofia/internal/1007%1.1.1.1"/> > >>> > >>> > >>> > >>> Still, Inbound is not hitting my FS console itself. Please assist where > >>> am > >>> I > >>> going wrong? > >>> > >>> Thanks, > >>> Rex > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> Brian West wrote: > >>>> > >>>> > >>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >>>> > >>>>> > >>>>> Hello, > >>>>> > >>>>> My public.xml configration is: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>> > >>>> $1 will not exist in this case because your regular expression doesn't > >>>> capture anything. So replace $1 with your target number or use > >>>> ^(123456)$ > >>>> > >>>> > >>>>> > >>>>> My default.xml configration is: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> data="sofia/sip/1001%freeswitchip"/> > >>>>> > >>>>> > >>>> > >>>> Can you elaborate how you're registering with your provider? > >>>> > >>>> > >>>>> > >>>>> > >>>>> When I am trying to call 123456 from my mobile no. Not able to see > any > >>>>> logging in FS console. Please assist where I am going wrong? Or do I > >>>>> require > >>>>> any extra modules to be installed? > >>>>> > >>>>> Thanks, > >>>>> Rex > >>>> > >>>> Brian West > >>>> brian at freeswitch.org > >>>> > >>>> -- Meet us at ClueCon! http://www.cluecon.com > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> > >>> -- > >>> View this message in context: > >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html > >>> Sent from the freeswitch-users mailing list archive at Nabble.com. > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -- > >> View this message in context: > >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html > >> Sent from the freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d45dc32931463593608! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/152e72bf/attachment-0002.html From talk2ram at gmail.com Tue Jun 9 00:02:41 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 12:32:41 +0530 Subject: [Freeswitch-users] mod_niible install problem Message-ID: Hi i have downloaded latest SVN and trying to make install i get the following error I googled for the same but there no information on this error how can i resolve this problem Ram making install mod_nibblebill Compiling mod_nibblebill.c... Compiling mod_nibblebill.c ... mod_nibblebill.c: In function ?get_balance?: mod_nibblebill.c:368: error: ?balance? undeclared (first use in this function) mod_nibblebill.c:368: error: (Each undeclared identifier is reported only once mod_nibblebill.c:368: error: for each function it appears in.) make[5]: *** [mod_nibblebill.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_nibblebill-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/89610d47/attachment-0002.html From talk2ram at gmail.com Tue Jun 9 00:29:30 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 12:59:30 +0530 Subject: [Freeswitch-users] I need a favor... In-Reply-To: <4A2B7580.9020909@gmail.com> References: <98a86adf0906062125h1d40e105md1b6ca36b56040c5@mail.gmail.com> <4A2B7580.9020909@gmail.com> Message-ID: On Sun, Jun 7, 2009 at 1:38 PM, Meftah Tayeb wrote: > hello, > welcome, i'm able to build a Installer for your Freeswitch > please: > i have a very small Internet connection (128KBPS) and FS Setup syse is 40MB > or +... > i give you only the setup project and you compile it... ok? > thanks > is this linux based or windows ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/32c9179d/attachment-0002.html From yivzhenko at mksat.net Tue Jun 9 01:26:03 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 9 Jun 2009 11:26:03 +0300 Subject: [Freeswitch-users] mod_nibblebill not set variable nibble_total_billed Message-ID: <200906091126.03556.yivzhenko@mksat.net> Some time ago mod_nibblebill was set variable nibble_total_billed after hangup. But after last few updates of module this variable is no more sets. Somebody else have this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9cf2cf56/attachment-0002.html From talk2ram at gmail.com Tue Jun 9 01:36:11 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 14:06:11 +0530 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: On Sun, May 31, 2009 at 8:39 PM, bakko wrote: > If you understand spanish please look at: > > http://www.freeswitch.es/node/55 > > Hi i have followed this URL iam using mysql, when i run the following command iam getting error what iam doing wrong ? python manage.py syncdb Traceback (most recent call last): File "manage.py", line 30, in from django.core.management import execute_manager ImportError: No module named django.core.management Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/5fdaa665/attachment-0002.html From durk.debeer at isp.solcon.nl Tue Jun 9 01:43:36 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Tue, 9 Jun 2009 10:43:36 +0200 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: References: Message-ID: Ok I've recieved an error message so if this message is being send a second time my deepest apologies for it. Hello Brian, I observed the problem by a Siemens Gigaset DE380 IPR and A Cisco 7960 and 7965. What happens is this, a call is coming in on Freeswitch and is being bridged to lets say the Siemens on extension 100. Now the person accepting the call on extension 100 wants to transfer the call to an other extension lets say 200. The normal scenario would be extension 100 puts the original call on hold, Freeswitch streams moh to the original call, extension 100 dials 200, the having a conversation, 100 transfers the original call too 200. Now here the problem begins. Normally extension 100 would send a refer sip message to freeswitch, who would then connect the original call to extension 200. The Siemens and Cisco phones do not send a refer sip message first. What the do is putting extension 200 on hold by means of an send only sip message. When Freeswitch is receiving this it streams moh to extension 200. After this the phones are sending the transfer by means of a sip refer message. When Freeswitch is receiving this it can't perform this transfer, that of original call to extension 200, because extension 100 has put extension 200 on receive only and extension 200 is receiving the moh. Resulting in an original call receiving moh and an extension 200 receiving moh. When this situation arises there's no way in connecting these to together. So what I need is a way to detect that there is an transfer by means of an sip refer message to a extension that has being put on hold. If so I need to get freeswitch to break this hold and transfer the original call to this extension. I hope that this will make the problem a little bit clearer. Kind regards Durk > This makes no sense.... Can you try to explain it more? I do attended > transfers with sip phones every day without a single problem. Maybe > i'm missing what you're talking about. > /b > On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote: >> Hello all, >> I have observed an issue on using Freeswitch and some SIP-phones. Ok >> the problem is this. Some phones, when attempting an attendant >> transfer, put the recipient of the transfer on hold. This results in >> Freeswitch starting to stream MOH music to the phone put on hold, if >> implemented. When now the original phone is pasing the transfer, >> Freeswitch is not going to process this transfer because the >> recipient end of it is on hold. It is however terminating the >> connections it has with the phone initiating the transfer. This >> means that the recipient of the transfer is never coming of hold >> again until it terminates the call. >> Is there a way to detect this behaviour, so I can get the recipient >> of hold before Freeswitch is processing the transfer?. >> Kind regards >> Durk > Brian West > brian at freeswitch.org From talk2ram at gmail.com Tue Jun 9 02:00:50 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 14:30:50 +0530 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: On Tue, Jun 9, 2009 at 2:06 PM, ram wrote: > > > On Sun, May 31, 2009 at 8:39 PM, bakko wrote: > >> If you understand spanish please look at: >> >> http://www.freeswitch.es/node/55 >> >> > > Hi > > i have followed this URL > > iam using mysql, > > when i run the following command iam getting error > > > what iam doing wrong ? > > python manage.py syncdb > Traceback (most recent call last): > File "manage.py", line 30, in > from django.core.management import execute_manager > ImportError: No module named django.core.management > > iam able to fix the above problem as per mentioned in the below URL http://forum.webfaction.com/viewtopic.php?id=324 now iam not able to start Freeswitch ./freeswitch 2009-06-09 01:57:45.826955 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-09 01:57:45.829038 [DEBUG] switch_event.c:552 Create event dispatch thread 0 Cannot Initialize [[error near line 972]: missing >] Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/b711a883/attachment-0002.html From mikael at bjerkeland.com Tue Jun 9 02:02:36 2009 From: mikael at bjerkeland.com (Mikael Aleksander Bjerkeland) Date: Tue, 09 Jun 2009 11:02:36 +0200 Subject: [Freeswitch-users] video transcoding In-Reply-To: <1244046640.28699.63.camel@localhost.localdomain> References: <1244046640.28699.63.camel@localhost.localdomain> Message-ID: <1244538156.4732.11.camel@mikael-xpsm1530> Hi, El mi?, 03-06-2009 a las 18:30 +0200, Francois Delawarde escribi?: > Hello, > > I'm interested in being able to do video transcoding mainly for > bridging 3G mobile and sip networks, and maybe later on some > conferencing with FS. > > Are video codecs planned to be added to FS even in a far future? Are > there copyright/patent problems with common video codecs (H.263 / > H.264) or with libraries (ffmpeg) that would prevent any of that from > happening? Yes, there are copyright/patent problems. The x264 library should be able to do transcoding but its license is not compatible with FS and some of the x264 developers are not interested in releasing it under a dual license. > > Meanwhile, would it be feasible to do some video transcoding using > external software (vlc?) with socket connections from-to FS? A guy I talked to in #x264 on FreeNode claims to be doing this, but not in a VoIP scenario. If you don't want to reinvent the wheel you probably have to create a wrapper for your socket interface to interface with x264. Please share your progress if you do so. If all else fails you could have a look at these to bridge the video call to PRI http://www.mirial.com/products/PSE_3G_Gateway.html http://www.radvision.com/Products/3GProductsApplications/SCOPIA3GVideoGateway/ And no, I do not know if they work :-) > > Thanks, > Fran?ois. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raffaele.p.guidi at gmail.com Tue Jun 9 02:14:58 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 9 Jun 2009 11:14:58 +0200 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: A side question: has anyone succesfully installed wikipbx on windows? On Tue, Jun 9, 2009 at 11:00, ram wrote: > > > On Tue, Jun 9, 2009 at 2:06 PM, ram wrote: > >> >> >> On Sun, May 31, 2009 at 8:39 PM, bakko wrote: >> >>> If you understand spanish please look at: >>> >>> http://www.freeswitch.es/node/55 >>> >>> >> >> Hi >> >> i have followed this URL >> >> iam using mysql, >> >> when i run the following command iam getting error >> >> >> what iam doing wrong ? >> >> python manage.py syncdb >> Traceback (most recent call last): >> File "manage.py", line 30, in >> from django.core.management import execute_manager >> ImportError: No module named django.core.management >> >> > > iam able to fix the above problem as per mentioned in the below URL > > http://forum.webfaction.com/viewtopic.php?id=324 > now iam not able to start Freeswitch > > ./freeswitch > 2009-06-09 01:57:45.826955 [INFO] switch_event.c:564 Activate Eventing > Engine. > 2009-06-09 01:57:45.829038 [DEBUG] switch_event.c:552 Create event dispatch > thread 0 > Cannot Initialize [[error near line 972]: missing >] > > > Ram > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/6f1664e8/attachment-0002.html From asannucci at gmail.com Tue Jun 9 02:41:35 2009 From: asannucci at gmail.com (bakko) Date: Tue, 9 Jun 2009 11:41:35 +0200 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com><9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com><99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: Look at /usr/local/freeswitch/log/freeswitch.xml.fsxml Line 972 This is a sintaxys error in the configuration. Chao -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/f8e08312/attachment-0002.html From talk2ram at gmail.com Tue Jun 9 02:46:32 2009 From: talk2ram at gmail.com (ram) Date: Tue, 9 Jun 2009 15:16:32 +0530 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> Message-ID: On Tue, Jun 9, 2009 at 3:11 PM, bakko wrote: > Look at /usr/local/freeswitch/log/freeswitch.xml.fsxml > > Line 972 > > This is a sintaxys error in the configuration. > > it was manual mistake when i edited that file fixed problem Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8f2c7677/attachment-0002.html From durk.debeer at isp.solcon.nl Tue Jun 9 02:55:24 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Tue, 9 Jun 2009 11:55:24 +0200 Subject: [Freeswitch-users] Problem with attendant transfer In-Reply-To: References: Message-ID: <1D1333C8C667434989D600F21BFFFC5E@solcon.local> Hallo Ray Siemens type Gigaset DE380 IPR, Cisco type 7965 and 7960. Also tested Grandstream type 2010 and Linksys type 921 no problem with these phones. I've downloaded a new firmware for the Siemens but wasn't able to test it jet. Durk > Durk de Beer wrote: >> >> Hello all, >> >> I have observed an issue on using Freeswitch and some SIP-phones. Ok >> the problem is this. Some phones >> > which phones? > -Ray From klaus.teller at gmx.net Tue Jun 9 05:17:10 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 09 Jun 2009 14:17:10 +0200 Subject: [Freeswitch-users] Reject call without answering Message-ID: <20090609121710.157260@gmx.net> Hi, I'm still looking for a way to reject a call without answering. I've tried various things without solution. >From the socket interface i tried: SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 call-command: execute execute-app-name: respond execute-app-arg: 503 >From Javascript, i tried each of the followings: session.hangup(); session.reject(); session.execute("respond", data="503"); In the two first cases as well as in the socket interface case, it seemed as if Freeswitch sent a kill signal to the VoiP provider. But that isn't enough apparently to cancel the call. In the third case, i get the message: Session is not active. Any other suggestion? Thanks, Klaus. -- GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 Euro/mtl.! http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a From rdenert at tng.de Tue Jun 9 05:56:49 2009 From: rdenert at tng.de (Rudolf Denert) Date: Tue, 9 Jun 2009 14:56:49 +0200 (CEST) Subject: [Freeswitch-users] DTMF Problems In-Reply-To: <5346021.351571244551867400.JavaMail.root@zimbra.tng.de> Message-ID: <11031925.351641244552209113.JavaMail.root@zimbra.tng.de> Hello, I can give the all-clear! It was my mistake ( ...and it was a silly one :-/ ) I had to applications that interfere each other. They are: I don't know why I skip that in my dialplan!!! http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Yes! A small mistake with a huge effect. But thanks for all you help. :-) Greetz ----- Urspr?ngliche Mail ----- Von: "Brian West" An: freeswitch-users at lists.freeswitch.org Gesendet: Montag, 8. Juni 2009 17:12:35 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] DTMF Problems I wrote that to demonstrate that exact situation but you still can't tell if they are inband or info :P /b On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote: Rudolf, I believe there is a snippet in the sample XML dialplan to detect the lack of telephone-event in the SDP and activate inband detection. You could use that for inspiration. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Jun 9 06:45:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 08:45:02 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <006101c9e8ae$11f03c00$35d0b400$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> Message-ID: You have a upnp device handing out 0.0.0.0 as the gateway address ... I'll patch that shortly to disable that. /b On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote: > http://pastebin.freeswitch.org/9319 > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Monday, June 08, 2009 7:39 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can't hear outbound calls > > pastebin your profile config and the output of global_getvar > > /b > > On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: > > > I had a working FS installation which I messed up by doing a fresh > install. I tried to integrate all my custom changes, but I?m sure I > screwed something up. > > The symptom is on an outbound call, sometimes I can hear ringing, > other times I cannot. Finally I can see FS connects via a softphone, > but I hear only silence. The other side of the conversation hears > static. > > I pasted a siptrace of the external profile. The Contact, Via and > SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 > > FS exists on a LAN behind a NAT firewall along with all its clients. > There is a SwitchVox system which predates the FS. I had to use an > external sip = 5090 for FS. Also I think I had to use a different > WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) > than the one FS came up with (xxx.xxx.xxx.82), but I can?t figure > out where I set this address. > > I would appreciate any help. Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/6e8853c6/attachment-0002.html From rex.alex345 at yahoo.com Tue Jun 9 06:46:06 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 9 Jun 2009 06:46:06 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> <1244480676920-3044219.post@n2.nabble.com> Message-ID: <1244555166651-3049491.post@n2.nabble.com> Incoming calls are not even hitting my FS box console. Peter Olsson wrote: > > Are you able to see anything at all in the console/log? > > I'm starting to doubt that the call even gets into the FS box... :) > > Try enabling more logs (console loglevel debug), and try again. > > > On 09-06-08 19.04, "Rex_Alex" wrote: > > > > > > Hello, > > Yes you are right. they are the on the same LAN. Inboun-acl added > in internal profile as well and restarted the FS completely. But no luck.. > Please help us to resolve the same.. > > Thanks, > Rex > > > Peter Olsson wrote: >> >> The PRI/SIP-box probably talks to the internal profile (I guess that they >> are on the same LAN). Try to add the inbound-acl to the internal profile >> as well. Also restart FS completely, just to be 100% sure that config is >> reloaded. >> >> //Peter >> >> >> On 09-06-08 18.34, "Rex_Alex" wrote: >> >> >> >> Hello, >> >> I am not sure about the profile which I am calling into. >> >> My scenario is this, I am trying to reach extn 1007 registered in FS >> server >> from my mobile through an inbound PRI connected to the audiocode with DID >> 123456. >> >> Thanks, >> Rex >> >> >> >> Peter Olsson wrote: >>> >>> I don't see what you've added. But I guess it's something like . >>> >>> Are you sure you're dialing into the external profile? It's on port 5080 >>> by default, and the internal is on 5060. >>> >>> /Peter >>> >>> >>> On 09-06-08 17.53, "Rex_Alex" wrote: >>> >>> >>> >>> Hello Peter, >>> >>> Yes, I have added >>> >>> >>> >>> under tag in sip_profiles/external.xml >>> >>> Thanks, >>> Rex >>> >>> >>> >>> Peter Olsson wrote: >>>> >>>> Have you configured the sip profile to use the acl list you have >>>> created >>>> (Inbound_Test)? >>>> >>>> /Peter >>>> >>>> >>>> ----- Ursprungligt meddelande ----- >>>> Fr?n: Rex_Alex >>>> Skickat: den 8 juni 2009 17:40 >>>> Till: freeswitch-users at lists.freeswitch.org >>>> >>>> ?mne: Re: [Freeswitch-users] Inbound using FS >>>> >>>> >>>> Hi Brian, >>>> >>>> I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 >>>> >>>> Addions made are, >>>> >>>> acl.conf.xml >>>> >>>> >>>> >>>> >>>> >>>> freeswitch.xml >>>> >>>> >>>> >>>> sip_profiles/external.xml >>>> >>>> under tag >>>> >>>> public.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> default.xml >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/internal/1007%1.1.1.1"/> >>>> >>>> >>>> >>>> Still, Inbound is not hitting my FS console itself. Please assist where >>>> am >>>> I >>>> going wrong? >>>> >>>> Thanks, >>>> Rex >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Brian West wrote: >>>>> >>>>> >>>>> On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: >>>>> >>>>>> >>>>>> Hello, >>>>>> >>>>>> My public.xml configration is: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> $1 will not exist in this case because your regular expression doesn't >>>>> capture anything. So replace $1 with your target number or use >>>>> ^(123456)$ >>>>> >>>>> >>>>>> >>>>>> My default.xml configration is: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/sip/1001%freeswitchip"/> >>>>>> >>>>>> >>>>> >>>>> Can you elaborate how you're registering with your provider? >>>>> >>>>> >>>>>> >>>>>> >>>>>> When I am trying to call 123456 from my mobile no. Not able to see >>>>>> any >>>>>> logging in FS console. Please assist where I am going wrong? Or do I >>>>>> require >>>>>> any extra modules to be installed? >>>>>> >>>>>> Thanks, >>>>>> Rex >>>>> >>>>> Brian West >>>>> brian at freeswitch.org >>>>> >>>>> -- Meet us at ClueCon! http://www.cluecon.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html >>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://n2.nabble.com/Inbound-using-FS-tp3012286p3043804.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://n2.nabble.com/Inbound-using-FS-tp3012286p3044052.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://n2.nabble.com/Inbound-using-FS-tp3012286p3044219.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4a2d45dc32931463593608! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3049491.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mrene_lists at avgs.ca Tue Jun 9 06:50:30 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 9 Jun 2009 09:50:30 -0400 Subject: [Freeswitch-users] mod_niible install problem In-Reply-To: References: Message-ID: <32EACEF7-594E-4332-8E6A-4230E76C79FA@avgs.ca> Hi, Install unixodbc-dev and run configure again. Math On 9-Jun-09, at 3:02 AM, ram wrote: > > Hi > > i have downloaded latest SVN > > and trying to make install > > i get the following error > > I googled for the same > but there no information on this error > > > how can i resolve this problem > > Ram > > making install mod_nibblebill > Compiling mod_nibblebill.c... > Compiling mod_nibblebill.c ... > mod_nibblebill.c: In function ?get_balance?: > mod_nibblebill.c:368: error: ?balance? undeclared (first use in this > function) > mod_nibblebill.c:368: error: (Each undeclared identifier is reported > only once > mod_nibblebill.c:368: error: for each function it appears in.) > make[5]: *** [mod_nibblebill.lo] Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_nibblebill-install] Error 1 > make[2]: *** [install-recursive] Error 1 > Making install in build > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rex.alex345 at yahoo.com Tue Jun 9 06:55:19 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Tue, 9 Jun 2009 06:55:19 -0700 (PDT) Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> <1244480676920-3044219.post@n2.nabble.com> <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> Message-ID: <1244555719062-3049537.post@n2.nabble.com> Hello, Below are some changes I have made, Post me if any additions required... acl.conf.xml freeswitch.xml sip_profiles/internal.xml < param name="apply-inbound-acl" value="inbound_ac" /> under tag public.xml default.xml Thanks Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3049537.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Tue Jun 9 07:26:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 9 Jun 2009 16:26:28 +0200 Subject: [Freeswitch-users] Inbound using FS In-Reply-To: <1244555719062-3049537.post@n2.nabble.com> References: <1243952053200-3012286.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E19B@cooper> <1244476410874-3043804.post@n2.nabble.com> <1244478876050-3044052.post@n2.nabble.com> <1244480676920-3044219.post@n2.nabble.com> <2ea4d47e0906082147y74a52d16iaad2981f1be6deb6@mail.gmail.com> <1244555719062-3049537.post@n2.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB8A7C@cooper> If you don't even see it when debug logging is enabled, there is something wrong in the other end. About the IP's. I guess you're just faking IP's in these email,s or are you using 2.2.2.2 and 1.1.1.1 for real? Cause in that case you're in trouble. I just wanted to make sure... :) /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Rex_Alex Skickat: den 9 juni 2009 15:55 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Inbound using FS Hello, Below are some changes I have made, Post me if any additions required... acl.conf.xml freeswitch.xml sip_profiles/internal.xml < param name="apply-inbound-acl" value="inbound_ac" /> under tag public.xml default.xml Thanks Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3049537.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a2e6b2232934656730255! From brian at freeswitch.org Tue Jun 9 07:41:14 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 09:41:14 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <006101c9e8ae$11f03c00$35d0b400$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> Message-ID: <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> update to 13719, rupa did a patch that fixed this.. also find that printer that gives out the 0.0.0.0 addr and turn off upnp :P /b On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote: > http://pastebin.freeswitch.org/9319 > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Monday, June 08, 2009 7:39 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can't hear outbound calls > > pastebin your profile config and the output of global_getvar > > /b > > On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: > > > I had a working FS installation which I messed up by doing a fresh > install. I tried to integrate all my custom changes, but I?m sure I > screwed something up. > > The symptom is on an outbound call, sometimes I can hear ringing, > other times I cannot. Finally I can see FS connects via a softphone, > but I hear only silence. The other side of the conversation hears > static. > > I pasted a siptrace of the external profile. The Contact, Via and > SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 > > FS exists on a LAN behind a NAT firewall along with all its clients. > There is a SwitchVox system which predates the FS. I had to use an > external sip = 5090 for FS. Also I think I had to use a different > WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) > than the one FS came up with (xxx.xxx.xxx.82), but I can?t figure > out where I set this address. > > I would appreciate any help. Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/1fdda4fa/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Jun 9 08:09:40 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 9 Jun 2009 16:09:40 +0100 Subject: [Freeswitch-users] mod vmd and lua - Solved Message-ID: I finally got around to looking at why mod vmd didn't appear to run when using LUA. Turned out that the example in the wiki was wrong. It should have been session:execute("vmd","start"); And not session:execute("vmd"); I've updated the wiki Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8f8932b2/attachment-0002.html From msc at freeswitch.org Tue Jun 9 08:44:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 08:44:24 -0700 Subject: [Freeswitch-users] ClueCon 2009 Information - Roommates Message-ID: <87f2f3b90906090844u4797759ew170f470cd19e0330@mail.gmail.com> FYI, We've had several people inquire about finding someone with whom to share a room for ClueCon. Please keep in mind that we do have a minimum number of rooms we need to book with the Wyndham. However, we would hate for someone not to attend simply because they could not afford a hotel room. To that end I would like to ask for a volunteer from the community to be the go-to person for helping roommates to get connected. Perhaps we could get more total rooms booked by helping those with rooming needs and who might not otherwise be able to come to ClueCon this year. Please email me off list if you are able to help. Thanks! -Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/6bc2a7fb/attachment-0002.html From msc at freeswitch.org Tue Jun 9 08:45:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 08:45:25 -0700 Subject: [Freeswitch-users] mod vmd and lua - Solved In-Reply-To: References: Message-ID: <87f2f3b90906090845n5e00e8d4r1436d55aa69f6a5@mail.gmail.com> > > > > I?ve updated the wiki > > > > You're a gentleman and a scholar! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/78a565ca/attachment-0002.html From larclap at yahoo.com Tue Jun 9 08:54:41 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 08:54:41 -0700 Subject: [Freeswitch-users] Error making FS configure Message-ID: <012001c9e91a$99c4c1c0$cd4e4540$@com> I tried to make FS current with the following command, which got errors in the ./configure step. Is it OK to proceed to make install? make clean && svn up && ./configure configure: creating ./config.status config.status: creating Makefile config.status: creating libedit.pc config.status: creating src/Makefile config.status: creating doc/Makefile config.status: creating examples/Makefile config.status: creating config.h config.status: executing depfiles commands configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/76dbc731/attachment-0002.html From gmaruzz at celliax.org Tue Jun 9 09:00:31 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 9 Jun 2009 18:00:31 +0200 Subject: [Freeswitch-users] broken compilation on windows? Message-ID: <7b197bef0906090900v6052e150vcff69406135d5645@mail.gmail.com> Hi all, I cannot compile on Windows the current svn, 13722. The first error it gives is: freeswitch\libs\pcre\pcre_internal.h(368) : fatal error C1189: #error : LINK_SIZE must be either 2, 3, or 4 then it fails 81 projects (42 succeeded), because no freeswitchcore.lib (obviously) I tried both the Freeswitch.2008.sln and the freeswitch.express.2008.sln, I'm using VC Express 2008. -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From krice at freeswitch.org Tue Jun 9 09:01:28 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 09 Jun 2009 11:01:28 -0500 Subject: [Freeswitch-users] Error making FS configure In-Reply-To: <012001c9e91a$99c4c1c0$cd4e4540$@com> Message-ID: Re-bootstrap if you just updated several fixes to the build system and an update to the pcre lab were recently commited... If you follow the ?trunk mailing everyone tries to tag commits that require this in their commit messages From: Lars Zeb Reply-To: Date: Tue, 9 Jun 2009 08:54:41 -0700 To: Subject: [Freeswitch-users] Error making FS configure I tried to make FS current with the following command, which got errors in the ../configure step. Is it OK to proceed to make install? make clean && svn up && ./configure configure: creating ./config.status config.status: creating Makefile config.status: creating libedit.pc config.status: creating src/Makefile config.status: creating doc/Makefile config.status: creating examples/Makefile config.status: creating config.h config.status: executing depfiles commands configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/02ecc57b/attachment-0002.html From larclap at yahoo.com Tue Jun 9 09:03:20 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 09:03:20 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> Message-ID: <012501c9e91b$cf654880$6e2fd980$@com> Brian, I'm curious, how can you tell that a printer is giving out the 0.0.0.0 addr? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 7:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls update to 13719, rupa did a patch that fixed this.. also find that printer that gives out the 0.0.0.0 addr and turn off upnp :P /b On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote: http://pastebin.freeswitch.org/9319 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 08, 2009 7:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls pastebin your profile config and the output of global_getvar /b On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote: I had a working FS installation which I messed up by doing a fresh install. I tried to integrate all my custom changes, but I'm sure I screwed something up. The symptom is on an outbound call, sometimes I can hear ringing, other times I cannot. Finally I can see FS connects via a softphone, but I hear only silence. The other side of the conversation hears static. I pasted a siptrace of the external profile. The Contact, Via and SDP shows an address 0.0.0.0. http://pastebin.freeswitch.org/9318 FS exists on a LAN behind a NAT firewall along with all its clients. There is a SwitchVox system which predates the FS. I had to use an external sip = 5090 for FS. Also I think I had to use a different WAN address (xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came up with (xxx.xxx.xxx.82), but I can't figure out where I set this address. I would appreciate any help. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/292ed15a/attachment-0002.html From mike at jerris.com Tue Jun 9 09:06:03 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 9 Jun 2009 12:06:03 -0400 Subject: [Freeswitch-users] Error making FS configure In-Reply-To: References: Message-ID: <4384608B-DEEE-4ADD-B532-5756077FBF6C@jerris.com> I updated the pcre lib last night, and it is trying to add files that are already there in your working copy. You can try doing: rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make current Mike On Jun 9, 2009, at 12:01 PM, Ken Rice wrote: > Re-bootstrap if you just updated several fixes to the build system > and an update to the pcre lab were recently commited... If you > follow the ?trunk mailing everyone tries to tag commits that require > this in their commit messages > > > From: Lars Zeb > Reply-To: > Date: Tue, 9 Jun 2009 08:54:41 -0700 > To: > Subject: [Freeswitch-users] Error making FS configure > > I tried to make FS current with the following command, which got > errors in the ../configure step. Is it OK to proceed to make install? > > make clean && svn up && ./configure > configure: creating ./config.status > config.status: creating Makefile > config.status: creating libedit.pc > config.status: creating src/Makefile > config.status: creating doc/Makefile > config.status: creating examples/Makefile > config.status: creating config.h > config.status: executing depfiles commands > configure: configuring in libs/pcre > configure: running /bin/sh './configure.gnu' --prefix=/usr/local/ > freeswitch --cache-file=/dev/null --srcdir=. > configure: error: cannot find sources (pcre.h.in) in . > configure: error: /bin/sh './configure.gnu' failed for libs/pcre > > Thanks, Lars > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/17edbe46/attachment-0002.html From brian at freeswitch.org Tue Jun 9 09:06:09 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 11:06:09 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <012501c9e91b$cf654880$6e2fd980$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> Message-ID: Because rupa on IRC is having the same problem.. Check the error message I print now and the device url will be printed thanks to rupa's patch. /b On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: > Brian, I?m curious, how can you tell that a printer is giving out > the 0.0.0.0 addr? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/e5f27091/attachment-0002.html From eman at chabotel.com Tue Jun 9 09:14:08 2009 From: eman at chabotel.com (freeswitch list) Date: Tue, 9 Jun 2009 12:14:08 -0400 Subject: [Freeswitch-users] Few questions Message-ID: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> 1) How do you setup a gateway where the provider doesn't require a user name and password? For freeswitch gateways username and password are mandatory. 2) Is there anyway to have xml_curl send the password for directory entry requests. 3) What is the best way to monitor for failed sip register attempts? 4) Is there a way to increase the volume of voicemail message sent by email? They are fine in freeswitch but when sent by email as mp3 (mod_shout) they are really low. 5) Is there any future plans to make the voicemail module more customizable? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/fac939e2/attachment-0002.html From msc at freeswitch.org Tue Jun 9 09:36:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 09:36:20 -0700 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> Message-ID: <87f2f3b90906090936t21d0919k717e553fa3e18a6d@mail.gmail.com> On Tue, Jun 9, 2009 at 9:14 AM, freeswitch list wrote: > 1) How do you setup a gateway where the provider doesn't require a user > name and password? For freeswitch gateways username and password are > mandatory. > Find this line in example.xml: Uncomment that line in your gateway and it won't attempt to register. Put a bogus username and password in those fields to keep FS from complaining and you're all set. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/73ef0f74/attachment-0002.html From brian at freeswitch.org Tue Jun 9 09:43:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 11:43:16 -0500 Subject: [Freeswitch-users] Few questions In-Reply-To: <87f2f3b90906090936t21d0919k717e553fa3e18a6d@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <87f2f3b90906090936t21d0919k717e553fa3e18a6d@mail.gmail.com> Message-ID: <6F01F1FD-09F0-4563-A439-68A404089CB4@freeswitch.org> But the bigger issue is if the far end doesn't require register and won't 401/407 you and does auth via IP you don't need a gateway. /b On Jun 9, 2009, at 11:36 AM, Michael Collins wrote: > > > On Tue, Jun 9, 2009 at 9:14 AM, freeswitch list > wrote: > 1) How do you setup a gateway where the provider doesn't require a > user name and password? For freeswitch gateways username and > password are mandatory. > > Find this line in example.xml: > > > Uncomment that line in your gateway and it won't attempt to > register. Put a bogus username and password in those fields to keep > FS from complaining and you're all set. > -MC Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/786a471c/attachment-0002.html From rupa at rupa.com Tue Jun 9 09:44:04 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 11:44:04 -0500 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> Message-ID: 2009/6/9 freeswitch list > 1) How do you setup a gateway where the provider doesn't require a user > name and password? For freeswitch gateways username and password are > mandatory. > Use the param: > > 2) Is there anyway to have xml_curl send the password for directory entry > requests. > dunno > > 3) What is the best way to monitor for failed sip register attempts? > You should get an event over event sockets on registration failure. You can also watch the log file for registration failure. Finally, "sofia status" will give you the current status of all gateways. You should also enable ping if you want to fail faster after the gateway becomes unreachable. > > 4) Is there a way to increase the volume of voicemail message sent by > email? They are fine in freeswitch but when sent by email as mp3 > (mod_shout) they are really low. > Look in shout.conf.xml -- there is a volume and outscale parameter. I haven't looked at the source to see exactly what they are for, but that should get you started. > > 5) Is there any future plans to make the voicemail module more > customizable? > If you can code -- patches welcome. Otherwise, you can file a request on jira with what you want. Add a bounty to encourage it to get done sooner rather than later. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/526d748a/attachment-0002.html From gmaruzz at celliax.org Tue Jun 9 09:45:59 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 9 Jun 2009 18:45:59 +0200 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems Message-ID: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From dujinfang at gmail.com Tue Jun 9 09:48:52 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 00:48:52 +0800 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> Message-ID: <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> > 2) Is there anyway to have xml_curl send the password for directory > entry requests. > try this: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7b8cc95c/attachment-0002.html From larclap at yahoo.com Tue Jun 9 10:00:04 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 10:00:04 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> Message-ID: <015a01c9e923$bf458020$3dd08060$@com> Thanks, Brian and Mike J and Ken R and Jason W. Outbound calls are now working OK. Brian, I don't know where to look for the 0.0.0.0 addr error message. I checked the log/freeswitch.log but did not recognize anything. I also noticed that nat_public_addr is not longer displayed in the global_getvar command. How is this value set? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 9:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls Because rupa on IRC is having the same problem.. Check the error message I print now and the device url will be printed thanks to rupa's patch. /b On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: Brian, I'm curious, how can you tell that a printer is giving out the 0.0.0.0 addr? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/de40e2ee/attachment-0002.html From brian at freeswitch.org Tue Jun 9 10:06:29 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 12:06:29 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015a01c9e923$bf458020$3dd08060$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> Message-ID: <6497F66C-4E87-412E-B222-7B6DEA789392@freeswitch.org> because 0.0.0.0 causes it to disable upnp cuz its invalid. /b On Jun 9, 2009, at 12:00 PM, Lars Zeb wrote: > Thanks, Brian and Mike J and Ken R and Jason W. > > Outbound calls are now working OK. > > Brian, I don?t know where to look for the 0.0.0.0 addr error > message. I checked the log/freeswitch.log but did not recognize > anything. > > I also noticed that nat_public_addr is not longer displayed in the > global_getvar command. How is this value set? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/507a9dbd/attachment-0002.html From teqx at yahoo.com Tue Jun 9 10:14:49 2009 From: teqx at yahoo.com (zigurds) Date: Tue, 9 Jun 2009 10:14:49 -0700 (PDT) Subject: [Freeswitch-users] bug in channel event other leg parameters? Message-ID: <23947374.post@talk.nabble.com> Hi, all. While expecting channel events that are fired in call lifetime, have encountered that for lot of channel events Other-Leg-Username points to user that belongs to channel event for what is fired. For example, let's say I'm calling from 101 to 102, when receiving events CHANNEL_UNBRIDGE, CHANNEL_EXECUTE_COMPLETE, CHANNEL_HANGUP, if channel name is Channel-Name: sofia/Test/101%40XXXX, I'm getting also Other-Leg-Username: 101. I was expected that there should be 102 or I'm misunderstanding something? Some other information in Other-Leg too releates to user 101. By the way, does there exist more detailed information about call events flow and what event attributes mean. Seems that http://wiki.freeswitch.org/wiki/Event_list is very uncomplete... Thanks, Zigurds -- View this message in context: http://www.nabble.com/bug-in-channel-event-other-leg-parameters--tp23947374p23947374.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From eman at chabotel.com Tue Jun 9 10:26:53 2009 From: eman at chabotel.com (freeswitch list) Date: Tue, 9 Jun 2009 13:26:53 -0400 Subject: [Freeswitch-users] Few questions In-Reply-To: <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> Message-ID: <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> Awesome. Thank you so much guys. One last question. How would I forward a call in my dialplan to my cell phone? I tried but the caller id changes to the gateway callerid. On Tue, Jun 9, 2009 at 12:48 PM, dujinfang wrote: > 2) Is there anyway to have xml_curl send the password for directory entry > requests. > > try this: > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/17b5eb8e/attachment-0002.html From rupa at rupa.com Tue Jun 9 11:15:47 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 13:15:47 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015a01c9e923$bf458020$3dd08060$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> Message-ID: If we get 0.0.0.0 as a public address, upnp support is disabled since we're getting the info from a gateway that can't route us to the internet. It is broken, so we don't trust it. The message you should see in the logs is something like: 2009-06-08 13:51:44.587812 [ERR] switch_nat.c:126 uPNP Device (url: http://192.168.1.2:4444/wipconn) returned an invalid external address of 0.0.0.0. Disabling uPNP 2009-06-08 13:51:44.587812 [INFO] switch_nat.c:380 No PMP or UPnP NAT detected! You'll probably not see this if you start fs in the background and then connect with fs_cli. So, look in your log files for it. The url will give you an idea as to which device is sending you invalid info. In my case it is a dlink router setup as a access point but still was implementing upnp (bad). I was able to disable upnp on that router. My printer (Epson Artisan 800) also participates in upnp, but it doesn't respond to the internet gateway stuff, so it was not the source of a problem for address discovery. It is causing me other issues but that is another story for another day for code that isn't committed yet. 2009/6/9 Lars Zeb > Thanks, Brian and Mike J and Ken R and Jason W. > > > > Outbound calls are now working OK. > > > > Brian, I don?t know where to look for the 0.0.0.0 addr error message. I > checked the log/freeswitch.log but did not recognize anything. > > > > I also noticed that nat_public_addr is not longer displayed in the > global_getvar command. How is this value set? > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Tuesday, June 09, 2009 9:06 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > Because rupa on IRC is having the same problem.. Check the error message I > print now and the device url will be printed thanks to rupa's patch. > > > > /b > > > > On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: > > > > Brian, I?m curious, how can you tell that a printer is giving out the > 0.0.0.0 addr? > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/19d8e8f1/attachment-0002.html From eman at chabotel.com Tue Jun 9 11:16:26 2009 From: eman at chabotel.com (freeswitch list) Date: Tue, 9 Jun 2009 14:16:26 -0400 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> Message-ID: <164a9ab00906091116gfef6046vec2362ef2d51c53a@mail.gmail.com> I couldn't get the gateway to work with register=false. I took the hint from Brian and scraped the gateway idea and put this in my dialplan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/123c05a5/attachment-0002.html From freeswitch-users at digitaldan.com Tue Jun 9 13:28:09 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Tue, 9 Jun 2009 14:28:09 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <25563227.12121244578744238.JavaMail.daniel@radio> Message-ID: <32308728.12141244579283168.JavaMail.daniel@radio> Hi guys, I'm noticing that when recording calls (using session.recordFile in both lua and javascript) the call is disconnecting after 12 hours (exactly 12 hours). I'm still looking into our cisco gateways to see if they are doing the call clearing, but wanted to know if there were any timers on calls or specically on recording calls. I'm using svn rev 13471 on debian, the calls originate from a cisco 53xx media gateway and are using sip / g.711 ulaw. Thanks. Dan- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/ab9ce5fd/attachment-0002.html From brian at freeswitch.org Tue Jun 9 13:39:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 15:39:30 -0500 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <32308728.12141244579283168.JavaMail.daniel@radio> References: <32308728.12141244579283168.JavaMail.daniel@radio> Message-ID: how big is the file? /b On Jun 9, 2009, at 3:28 PM, freeswitch-users at digitaldan.com wrote: > Hi guys, > > I'm noticing that when recording calls (using session.recordFile in > both lua and javascript) the call is disconnecting after 12 hours > (exactly 12 hours). I'm still looking into our cisco gateways to > see if they are doing the call clearing, but wanted to know if > there were any timers on calls or specically on recording calls. > I'm using svn rev 13471 on debian, the calls originate from a cisco > 53xx media gateway and are using sip / g.711 ulaw. > > Thanks. > Dan- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/d3b3d609/attachment-0002.html From freeswitch-users at digitaldan.com Tue Jun 9 13:57:09 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 9 Jun 2009 14:57:09 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: Message-ID: <32268694.12161244581025121.JavaMail.daniel@radio> 330M (345723426 bytes) D- ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 9, 2009 2:39:30 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording time limit? how big is the file? /b On Jun 9, 2009, at 3:28 PM, freeswitch-users at digitaldan.com wrote: Hi guys, I'm noticing that when recording calls (using session.recordFile in both lua and javascript) the call is disconnecting after 12 hours (exactly 12 hours). I'm still looking into our cisco gateways to see if they are doing the call clearing, but wanted to know if there were any timers on calls or specically on recording calls. I'm using svn rev 13471 on debian, the calls originate from a cisco 53xx media gateway and are using sip / g.711 ulaw. Thanks. Dan- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/800fc6bb/attachment-0002.html From larclap at yahoo.com Tue Jun 9 14:26:00 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 14:26:00 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> Message-ID: <020001c9e948$e2cca7b0$a865f710$@com> Rupa, Thanks for the detailed response. After upgrading from 13639 to 13732, I see no log errors. I am accessing Freeswitch vi fs_cli, but I did look in log/freeswitch.log. Certainly I see nothing that looks like your ERR below. I too have a dlink router. I will look at its configuration and see if upnp is enabled. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 11:16 AM To: freeswitch-users Subject: Re: [Freeswitch-users] Can't hear outbound calls If we get 0.0.0.0 as a public address, upnp support is disabled since we're getting the info from a gateway that can't route us to the internet. It is broken, so we don't trust it. The message you should see in the logs is something like: 2009-06-08 13:51:44.587812 [ERR] switch_nat.c:126 uPNP Device (url: http://192.168.1.2:4444/wipconn) returned an invalid external address of 0.0.0.0. Disabling uPNP 2009-06-08 13:51:44.587812 [INFO] switch_nat.c:380 No PMP or UPnP NAT detected! You'll probably not see this if you start fs in the background and then connect with fs_cli. So, look in your log files for it. The url will give you an idea as to which device is sending you invalid info. In my case it is a dlink router setup as a access point but still was implementing upnp (bad). I was able to disable upnp on that router. My printer (Epson Artisan 800) also participates in upnp, but it doesn't respond to the internet gateway stuff, so it was not the source of a problem for address discovery. It is causing me other issues but that is another story for another day for code that isn't committed yet. 2009/6/9 Lars Zeb Thanks, Brian and Mike J and Ken R and Jason W. Outbound calls are now working OK. Brian, I don't know where to look for the 0.0.0.0 addr error message. I checked the log/freeswitch.log but did not recognize anything. I also noticed that nat_public_addr is not longer displayed in the global_getvar command. How is this value set? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 9:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls Because rupa on IRC is having the same problem.. Check the error message I print now and the device url will be printed thanks to rupa's patch. /b On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote: Brian, I'm curious, how can you tell that a printer is giving out the 0.0.0.0 addr? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/81946ee9/attachment-0002.html From brian at freeswitch.org Tue Jun 9 14:32:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 16:32:30 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <020001c9e948$e2cca7b0$a865f710$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> Message-ID: You have to start freeswitch without -nc to see it. Only happens during start up. /b On Jun 9, 2009, at 4:26 PM, Lars Zeb wrote: > Rupa, > > Thanks for the detailed response. After upgrading from 13639 to > 13732, I see no log errors. I am accessing Freeswitch vi fs_cli, but > I did look in log/freeswitch.log. Certainly I see nothing that looks > like your ERR below. > > I too have a dlink router. I will look at its configuration and see > if upnp is enabled. > > Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9cadd1a0/attachment-0002.html From brian at freeswitch.org Tue Jun 9 14:36:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 16:36:26 -0500 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <32268694.12161244581025121.JavaMail.daniel@radio> References: <32268694.12161244581025121.JavaMail.daniel@radio> Message-ID: What format are you recording in ? And what is the hangup cause? Happen to have a sip trace? /b On Jun 9, 2009, at 3:57 PM, Dan wrote: > 330M (345723426 bytes) > > D- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/4012318d/attachment-0002.html From freeswitch-users at digitaldan.com Tue Jun 9 14:52:19 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 9 Jun 2009 15:52:19 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: Message-ID: <9615666.12391244584331944.JavaMail.daniel@radio> The freeswitch log show this before it hits my hangup hook in lua. I don't have a pcap trace of the sip messaging, I can try that tonight, it would definitely show where the hangup is coming from. I'm recording to a local disk as a .ul file, so a headerless ulaw format. 2009-06-08 21:25:44 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/external/5555551212 at 192.168.3.21 entering state [terminated][200] 2009-06-08 21:25:44 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/external/5555551212 at 192.168.3.21 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-08 21:25:44 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/external/5555551212 at 192.168.3.21 [KILL] 2009-06-08 21:25:44 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/external/5555551212 at 192.168.3.21 [BREAK] 2009-06-08 21:25:44 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read_codec() Restore original codec. 2009-06-08 21:25:44 [NOTICE] switch_cpp.cpp:1122 console_log() hangupHook: 272423 status: hangup Thanks D- ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 9, 2009 3:36:26 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording time limit? What format are you recording in ? And what is the hangup cause? Happen to have a sip trace? /b On Jun 9, 2009, at 3:57 PM, Dan wrote: 330M (345723426 bytes) D- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/929b03b8/attachment-0002.html From brian at freeswitch.org Tue Jun 9 14:55:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 16:55:51 -0500 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <9615666.12391244584331944.JavaMail.daniel@radio> References: <9615666.12391244584331944.JavaMail.daniel@radio> Message-ID: <08122ADB-8711-418E-B580-033ACEC9339E@freeswitch.org> Someone hung the call up thats all I can imagine. Its not going to be us but maybe the far side you're talking to. Look at the sip trace bet the far end is sending you a BYE /b On Jun 9, 2009, at 4:52 PM, Dan wrote: > The freeswitch log show this before it hits my hangup hook in lua. > I don't have a pcap trace of the sip messaging, I can try that > tonight, it would definitely show where the hangup is coming from. > I'm recording to a local disk as a .ul file, so a headerless ulaw > format. > > 2009-06-08 21:25:44 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() > Channel sofia/external/5555551212 at 192.168.3.21 entering state > [terminated][200] > 2009-06-08 21:25:44 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() > Hangup sofia/external/5555551212 at 192.168.3.21 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-06-08 21:25:44 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal sofia/external/5555551212 at 192.168.3.21 > [KILL] > 2009-06-08 21:25:44 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal sofia/external/5555551212 at 192.168.3.21 > [BREAK] > 2009-06-08 21:25:44 [DEBUG] switch_core_codec.c:122 > switch_core_session_set_read_codec() Restore original codec. > 2009-06-08 21:25:44 [NOTICE] switch_cpp.cpp:1122 console_log() > hangupHook: 272423 status: hangup > > > > Thanks > D- Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/80855e54/attachment-0002.html From juanma.v82 at gmail.com Tue Jun 9 15:04:18 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Tue, 9 Jun 2009 19:04:18 -0300 Subject: [Freeswitch-users] Set Codec per Endpoint Message-ID: Hello, Is it posible to set a codec-pref per Endpoint instead to set it in sip-profiles? In my directory: I do this but FS do codec negotiation with all codecs in the profile internal. What i am doing wrong? Thx in advance From msc at freeswitch.org Tue Jun 9 15:06:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 9 Jun 2009 15:06:01 -0700 Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <08122ADB-8711-418E-B580-033ACEC9339E@freeswitch.org> References: <9615666.12391244584331944.JavaMail.daniel@radio> <08122ADB-8711-418E-B580-033ACEC9339E@freeswitch.org> Message-ID: <87f2f3b90906091506s5da8b925u256eaeaaf8c0fbb3@mail.gmail.com> On Tue, Jun 9, 2009 at 2:55 PM, Brian West wrote: > Someone hung the call up thats all I can imagine. Its not going to be us > but maybe the far side you're talking to. Look at the sip trace bet the far > end is sending you a BYE > This would make sense. Some systems will have a hard limit on the length of a call and will disconnect automatically because they interpret a 12 hour call as a "problem" and the "solution" is to hang up. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/87f29783/attachment-0002.html From brian at freeswitch.org Tue Jun 9 15:10:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 17:10:07 -0500 Subject: [Freeswitch-users] Set Codec per Endpoint In-Reply-To: References: Message-ID: Do something like this Before you bridge /b On Jun 9, 2009, at 5:04 PM, JuanMa wrote: > codec-prefs Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/170d10ca/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 16:35:56 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 9 Jun 2009 19:35:56 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates Message-ID: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> Hi everyone, I'm trying to write a calling card app with A-Z rates, and I plan to use mod_lcr for this case... the only thing I need mod_lcr to do for me is get the rate based on the destination number / prefix. Is there a way I could achieve this with mod_lcr? I seen the wiki page and the SQL examples, but the SQL examples does a lot more, so I was thinking if I could use a custom SQL query to only do what I need. Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/e7a423e3/attachment-0002.html From larclap at yahoo.com Tue Jun 9 16:43:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 16:43:03 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <94240995-8A02-4997-9E9A-F9BB1AD7CABC@freeswitch.org> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> Message-ID: <024701c9e95c$07e71ea0$17b55be0$@com> Thanks for the explanation, Brian; it was lost on me before. It was a DLink DIR-625 which had UPnP enabled. I turned it off. It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. 2009-06-09 16:24:32.271913 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-09 16:24:32.274131 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-09 16:24:32.663627 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-09 16:24:32.664053 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-09 16:24:32.913583 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-09 16:24:32.914581 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-09 16:24:33.415479 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-09 16:24:34.415249 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-09 16:24:36.413782 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-09 16:24:38.906588 [ERR] switch_nat.c:96 uPNP Device (url: http://192.168.10.253:4444/wipconn) returned an invalid external address of 0.0.0.0. Disabling uPNP 2009-06-09 16:24:38.906633 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! 2009-06-09 16:24:38.908650 [INFO] switch_core_sqldb.c:507 Opening DB 2009-06-09 16:24:38.950200 [NOTICE] switch_scheduler.c:166 Starting task thread Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 2:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls You have to start freeswitch without -nc to see it. Only happens during start up. /b On Jun 9, 2009, at 4:26 PM, Lars Zeb wrote: Rupa, Thanks for the detailed response. After upgrading from 13639 to 13732, I see no log errors. I am accessing Freeswitch vi fs_cli, but I did look in log/freeswitch.log. Certainly I see nothing that looks like your ERR below. I too have a dlink router. I will look at its configuration and see if upnp is enabled. Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8bd8a80e/attachment-0002.html From rupa at rupa.com Tue Jun 9 17:18:50 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 19:18:50 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <024701c9e95c$07e71ea0$17b55be0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> Message-ID: On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9d7fb251/attachment-0002.html From larclap at yahoo.com Tue Jun 9 17:35:51 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 17:35:51 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <006101c9e8ae$11f03c00$35d0b400$@com> <6AB71346-DDD7-4F23-9556-3DA3D68B7438@freeswitch.org> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> Message-ID: <027501c9e963$69e8fa40$3dbaeec0$@com> Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/7b625e35/attachment-0002.html From rupa at rupa.com Tue Jun 9 17:37:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 19:37:24 -0500 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> Message-ID: Diego, Here is how I'd go about doing what I think you want. As payment, add a section to the wiki when you have it working. Create two profiles in lcr.conf. the first profile is your callingcard rate deck. Give it a profile id of 1. Then load your data into the lcr tables. carriers = define your carrier. call it whatever you want carrier_gateteway = you won't care about any real routes, so just load dummy data in here (linked to your carrier). lcr = load your rate deck here. Set profile id to 1. Now, to look up the customer's code, use the lcr application. application="lcr" data="$1 profilename" where profilename is the profile defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 format minus the + -- this is discussed in the wiki). At this point, you'll have the results of the lcr query in channel vars. ${lcr_route_count} tells you the number of routes found (if you loaded your rate deck right it should always be 1). ${lcr_rate_1} will contain the rate. So now you can set that to the var you need for nibblebill to work. If you want to use lcr to actually route the actual call, just call it again. This time with the profile id set to whatever you use to load the full lcr table for all your providers. On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: > Hi everyone, > > I'm trying to write a calling card app with A-Z rates, and I plan to use > mod_lcr for this case... the only thing I need mod_lcr to do for me is get > the rate based on the destination number / prefix. > > Is there a way I could achieve this with mod_lcr? I seen the wiki page and > the SQL examples, but the SQL examples does a lot more, so I was thinking if > I could use a custom SQL query to only do what I need. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/9e6e5e47/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 17:52:38 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 9 Jun 2009 20:52:38 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> Message-ID: <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> Thanks for your help Rupa :). Don't worry that I will give everything back to the wiki, as I learn more and more, I have also contributed back some things to the wiki: http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola I love to do that, I will also contribute my calling card application to the community as soon as I'm done with it ;). Regards, Diego On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: > Diego, > > Here is how I'd go about doing what I think you want. As payment, add a > section to the wiki when you have it working. > > Create two profiles in lcr.conf. > > the first profile is your callingcard rate deck. Give it a profile id of > 1. Then load your data into the lcr tables. > > carriers = define your carrier. call it whatever you want > carrier_gateteway = you won't care about any real routes, so just load > dummy data in here (linked to your carrier). > lcr = load your rate deck here. Set profile id to 1. > > Now, to look up the customer's code, use the lcr application. > > application="lcr" data="$1 profilename" where profilename is the profile > defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 > format minus the + -- this is discussed in the wiki). > > At this point, you'll have the results of the lcr query in channel vars. > > ${lcr_route_count} tells you the number of routes found (if you loaded your > rate deck right it should always be 1). ${lcr_rate_1} will contain the > rate. > > So now you can set that to the var you need for nibblebill to work. > > If you want to use lcr to actually route the actual call, just call it > again. This time with the profile id set to whatever you use to load the > full lcr table for all your providers. > > On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: > >> Hi everyone, >> >> I'm trying to write a calling card app with A-Z rates, and I plan to use >> mod_lcr for this case... the only thing I need mod_lcr to do for me is get >> the rate based on the destination number / prefix. >> >> Is there a way I could achieve this with mod_lcr? I seen the wiki page and >> the SQL examples, but the SQL examples does a lot more, so I was thinking if >> I could use a custom SQL query to only do what I need. >> >> Thanks, >> >> Diego >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/8ada7ec6/attachment-0002.html From rupa at rupa.com Tue Jun 9 18:05:29 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 9 Jun 2009 20:05:29 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <027501c9e963$69e8fa40$3dbaeec0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> Message-ID: if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: > Rupa, > > > > What options do I have for setting up logging? I?m sorry, but I don?t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 5:19 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > > > On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > > You might want to review how you have your logging setup then. The example > I gave you was copied/pasted out of my freeswitch.log file while testing > this fix. > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/3a070c79/attachment-0002.html From dujinfang at gmail.com Tue Jun 9 18:08:37 2009 From: dujinfang at gmail.com (seven) Date: Wed, 10 Jun 2009 09:08:37 +0800 Subject: [Freeswitch-users] Few questions In-Reply-To: <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> References: <164a9ab00906090914v76e04c9qa0276b39b16d0edb@mail.gmail.com> <7DF80F1F-25EB-4BDB-89F3-18ABFF06EEF9@gmail.com> <164a9ab00906091026r2540eb29yd94d637f66df7c75@mail.gmail.com> Message-ID: <3DE17717-F722-488F-84C7-AFC805EE4137@gmail.com> On Jun 10, 2009, at 1:26 AM, freeswitch list wrote: > Awesome. Thank you so much guys. > > One last question. How would I forward a call in my dialplan to my > cell phone? > > I tried > > > > > but the caller id changes to the gateway callerid. > depend on your gateway provider, remember FreeSWITCH is a B2BUA. If your provider doesn't allow custom caller id, there's no way to ... > > On Tue, Jun 9, 2009 at 12:48 PM, dujinfang > wrote: >> 2) Is there anyway to have xml_curl send the password for directory >> entry requests. >> > > try this: > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/b71b9d44/attachment-0002.html From larclap at yahoo.com Tue Jun 9 19:25:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 19:25:12 -0700 Subject: [Freeswitch-users] Documentation error in dialplan XML? Message-ID: <02a801c9e972$b07f0df0$117d29d0$@com> Is the closing of the condition element correct? I'm new at XML. Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/a9046edb/attachment-0002.html From brian at freeswitch.org Tue Jun 9 19:32:04 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 21:32:04 -0500 Subject: [Freeswitch-users] Documentation error in dialplan XML? In-Reply-To: <02a801c9e972$b07f0df0$117d29d0$@com> References: <02a801c9e972$b07f0df0$117d29d0$@com> Message-ID: Nope shouldn't be there .. if you can update the wiki that would be great. /b On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote: > Is the closing of the condition element correct? I?m new at XML. > > > > > > should the slash at the end of the element be there? --> > > > > > > Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/5d8a2106/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 20:53:11 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 9 Jun 2009 23:53:11 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> Message-ID: <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> Hi everyone, I have used freeswitch/scripts/contrib/intralanman/C/lcr/sql/mysql-5.0.sql to load the mod_lcr schema, that worked well. But whenever I try to insert data from the "Sample Data" in the wiki it fails: http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data mysql> INSERT INTO lcr (digits, rate, carrier_id, lead_strip, trail_strip, -> prefix, suffix, -> date_start, date_end, quality, reliability) VALUES -> ('1', 0.15, 1, 0, 0, '', '', -> current_timestamp - interval 1 year, -> current_timestamp + interval 1 year -> , 0, 0); ERROR 1452 (23000): Cannot add or update a child row: a foreign key constraint fails (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) mysql> Regards, Diego On Tue, Jun 9, 2009 at 8:52 PM, Diego Viola wrote: > Thanks for your help Rupa :). > > Don't worry that I will give everything back to the wiki, as I learn more > and more, I have also contributed back some things to the wiki: > > http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola > > I love to do that, I will also contribute my calling card application to > the community as soon as I'm done with it ;). > > Regards, > > Diego > > > On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: > >> Diego, >> >> Here is how I'd go about doing what I think you want. As payment, add a >> section to the wiki when you have it working. >> >> Create two profiles in lcr.conf. >> >> the first profile is your callingcard rate deck. Give it a profile id of >> 1. Then load your data into the lcr tables. >> >> carriers = define your carrier. call it whatever you want >> carrier_gateteway = you won't care about any real routes, so just load >> dummy data in here (linked to your carrier). >> lcr = load your rate deck here. Set profile id to 1. >> >> Now, to look up the customer's code, use the lcr application. >> >> application="lcr" data="$1 profilename" where profilename is the profile >> defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 >> format minus the + -- this is discussed in the wiki). >> >> At this point, you'll have the results of the lcr query in channel vars. >> >> ${lcr_route_count} tells you the number of routes found (if you loaded >> your rate deck right it should always be 1). ${lcr_rate_1} will contain the >> rate. >> >> So now you can set that to the var you need for nibblebill to work. >> >> If you want to use lcr to actually route the actual call, just call it >> again. This time with the profile id set to whatever you use to load the >> full lcr table for all your providers. >> >> On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: >> >>> Hi everyone, >>> >>> I'm trying to write a calling card app with A-Z rates, and I plan to use >>> mod_lcr for this case... the only thing I need mod_lcr to do for me is get >>> the rate based on the destination number / prefix. >>> >>> Is there a way I could achieve this with mod_lcr? I seen the wiki page >>> and the SQL examples, but the SQL examples does a lot more, so I was >>> thinking if I could use a custom SQL query to only do what I need. >>> >>> Thanks, >>> >>> Diego >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -Rupa >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/212ad3b1/attachment-0002.html From mrene_lists at avgs.ca Tue Jun 9 20:55:00 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 9 Jun 2009 23:55:00 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> Message-ID: I think a foreign key constraint is failing, should look into that On 9-Jun-09, at 11:53 PM, Diego Viola wrote: > Hi everyone, > > I have used freeswitch/scripts/contrib/intralanman/C/lcr/sql/ > mysql-5.0.sql to load the mod_lcr schema, that worked well. > > But whenever I try to insert data from the "Sample Data" in the wiki > it fails: http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data > > mysql> INSERT INTO lcr (digits, rate, carrier_id, lead_strip, > trail_strip, > -> prefix, suffix, > -> date_start, date_end, quality, reliability) > VALUES > -> ('1', 0.15, 1, 0, 0, '', '', > -> current_timestamp - interval 1 year, > -> current_timestamp + interval 1 year > -> , 0, 0); > ERROR 1452 (23000): Cannot add or update a child row: a foreign key > constraint fails (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY > (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON > UPDATE CASCADE) > mysql> > > Regards, > > Diego > > On Tue, Jun 9, 2009 at 8:52 PM, Diego Viola > wrote: > Thanks for your help Rupa :). > > Don't worry that I will give everything back to the wiki, as I learn > more and more, I have also contributed back some things to the wiki: > > http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola > > I love to do that, I will also contribute my calling card > application to the community as soon as I'm done with it ;). > > Regards, > > Diego > > > On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: > Diego, > > Here is how I'd go about doing what I think you want. As payment, > add a section to the wiki when you have it working. > > Create two profiles in lcr.conf. > > the first profile is your callingcard rate deck. Give it a profile > id of 1. Then load your data into the lcr tables. > > carriers = define your carrier. call it whatever you want > carrier_gateteway = you won't care about any real routes, so just > load dummy data in here (linked to your carrier). > lcr = load your rate deck here. Set profile id to 1. > > Now, to look up the customer's code, use the lcr application. > > application="lcr" data="$1 profilename" where profilename is the > profile defined in lcr.conf with id 1. $1 is the normalized number > (I suggest e164 format minus the + -- this is discussed in the wiki). > > At this point, you'll have the results of the lcr query in channel > vars. > > ${lcr_route_count} tells you the number of routes found (if you > loaded your rate deck right it should always be 1). ${lcr_rate_1} > will contain the rate. > > So now you can set that to the var you need for nibblebill to work. > > If you want to use lcr to actually route the actual call, just call > it again. This time with the profile id set to whatever you use to > load the full lcr table for all your providers. > > On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola > wrote: > Hi everyone, > > I'm trying to write a calling card app with A-Z rates, and I plan to > use mod_lcr for this case... the only thing I need mod_lcr to do for > me is get the rate based on the destination number / prefix. > > Is there a way I could achieve this with mod_lcr? I seen the wiki > page and the SQL examples, but the SQL examples does a lot more, so > I was thinking if I could use a custom SQL query to only do what I > need. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/baa3225e/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 21:13:38 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 00:13:38 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> Message-ID: <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> Any other ideas? On Tue, Jun 9, 2009 at 11:55 PM, Mathieu Rene wrote: > I think a foreign key constraint is failing, should look into that > > On 9-Jun-09, at 11:53 PM, Diego Viola wrote: > > Hi everyone, > > I have used freeswitch/scripts/contrib/intralanman/C/lcr/sql/mysql-5.0.sql > to load the mod_lcr schema, that worked well. > > But whenever I try to insert data from the "Sample Data" in the wiki it > fails: http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data > > mysql> INSERT INTO lcr (digits, rate, carrier_id, lead_strip, trail_strip, > -> prefix, suffix, > -> date_start, date_end, quality, reliability) VALUES > -> ('1', 0.15, 1, 0, 0, '', '', > -> current_timestamp - interval 1 year, > -> current_timestamp + interval 1 year > -> , 0, 0); > ERROR 1452 (23000): Cannot add or update a child row: a foreign key > constraint fails (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY > (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE > CASCADE) > mysql> > > Regards, > > Diego > > On Tue, Jun 9, 2009 at 8:52 PM, Diego Viola wrote: > >> Thanks for your help Rupa :). >> >> Don't worry that I will give everything back to the wiki, as I learn more >> and more, I have also contributed back some things to the wiki: >> >> http://wiki.freeswitch.org/wiki/Special:Contributions/Diego.viola >> >> I love to do that, I will also contribute my calling card application to >> the community as soon as I'm done with it ;). >> >> Regards, >> >> Diego >> >> >> On Tue, Jun 9, 2009 at 8:37 PM, Rupa Schomaker wrote: >> >>> Diego, >>> >>> Here is how I'd go about doing what I think you want. As payment, add a >>> section to the wiki when you have it working. >>> >>> Create two profiles in lcr.conf. >>> >>> the first profile is your callingcard rate deck. Give it a profile id of >>> 1. Then load your data into the lcr tables. >>> >>> carriers = define your carrier. call it whatever you want >>> carrier_gateteway = you won't care about any real routes, so just load >>> dummy data in here (linked to your carrier). >>> lcr = load your rate deck here. Set profile id to 1. >>> >>> Now, to look up the customer's code, use the lcr application. >>> >>> application="lcr" data="$1 profilename" where profilename is the profile >>> defined in lcr.conf with id 1. $1 is the normalized number (I suggest e164 >>> format minus the + -- this is discussed in the wiki). >>> >>> At this point, you'll have the results of the lcr query in channel vars. >>> >>> ${lcr_route_count} tells you the number of routes found (if you loaded >>> your rate deck right it should always be 1). ${lcr_rate_1} will contain the >>> rate. >>> >>> So now you can set that to the var you need for nibblebill to work. >>> >>> If you want to use lcr to actually route the actual call, just call it >>> again. This time with the profile id set to whatever you use to load the >>> full lcr table for all your providers. >>> >>> On Tue, Jun 9, 2009 at 6:35 PM, Diego Viola wrote: >>> >>>> Hi everyone, >>>> >>>> I'm trying to write a calling card app with A-Z rates, and I plan to use >>>> mod_lcr for this case... the only thing I need mod_lcr to do for me is get >>>> the rate based on the destination number / prefix. >>>> >>>> Is there a way I could achieve this with mod_lcr? I seen the wiki page >>>> and the SQL examples, but the SQL examples does a lot more, so I was >>>> thinking if I could use a custom SQL query to only do what I need. >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -Rupa >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/93fd3452/attachment-0002.html From larclap at yahoo.com Tue Jun 9 21:21:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 9 Jun 2009 21:21:12 -0700 Subject: [Freeswitch-users] Documentation error in dialplan XML? In-Reply-To: References: <02a801c9e972$b07f0df0$117d29d0$@com> Message-ID: <02b901c9e982$e4ea2420$aebe6c60$@com> Done From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, June 09, 2009 7:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Documentation error in dialplan XML? Nope shouldn't be there .. if you can update the wiki that would be great. /b On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote: Is the closing of the condition element correct? I'm new at XML. Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/d8d3a3a1/attachment-0002.html From mrene_lists at avgs.ca Tue Jun 9 21:21:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 10 Jun 2009 00:21:48 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> Message-ID: If we look at the message again, (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) And put it into words: The contrainst names fs_lcr/lcr affecting field carrier_id, and referencing the id field of the carriers table, fails. In other words, the value you have for carrier_id does not match any value of id in the carriers table. Math On 10-Jun-09, at 12:13 AM, Diego Viola wrote: > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) From brian at freeswitch.org Tue Jun 9 21:23:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 9 Jun 2009 23:23:15 -0500 Subject: [Freeswitch-users] Documentation error in dialplan XML? In-Reply-To: <02b901c9e982$e4ea2420$aebe6c60$@com> References: <02a801c9e972$b07f0df0$117d29d0$@com> <02b901c9e982$e4ea2420$aebe6c60$@com> Message-ID: <6428E0C1-1EF3-4219-8C51-18C8716D2105@freeswitch.org> Kewl, thanks! /b Sent from my iPhone On Jun 9, 2009, at 11:21 PM, "Lars Zeb" wrote: > Done > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: Tuesday, June 09, 2009 7:32 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Documentation error in dialplan XML? > > > > Nope shouldn't be there .. if you can update the wiki that would be > great. > > > > /b > > > > On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote: > > > > > Is the closing of the condition element correct? I?m new at XML. > > > > > > > > > > > > ?should the slash at the end of the element be there? --> > > > > > > > > > > > > Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090609/5a50a5c2/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 21:43:35 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 00:43:35 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> Message-ID: <86a32abc0906092143yb0262ddpd737c4fc45b9fa3c@mail.gmail.com> Fixed the issue, I will update the wiki now with a MySQL example. Regards, Diego On Wed, Jun 10, 2009 at 12:21 AM, Mathieu Rene wrote: > If we look at the message again, > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) > And put it into words: > The contrainst names fs_lcr/lcr affecting field carrier_id, and > referencing the id field of the carriers table, fails. > In other words, the value you have for carrier_id does not match any > value of id in the carriers table. > > Math > > On 10-Jun-09, at 12:13 AM, Diego Viola wrote: > > > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) > > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/10d23e68/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 22:07:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 01:07:47 -0400 Subject: [Freeswitch-users] mod_lcr for a-z rates In-Reply-To: <86a32abc0906092143yb0262ddpd737c4fc45b9fa3c@mail.gmail.com> References: <86a32abc0906091635u37afdfcm33966ccea64266b4@mail.gmail.com> <86a32abc0906091752n69a19df8k9323ce5e0f33c72f@mail.gmail.com> <86a32abc0906092053s1555b898o35bdff94fc2d9392@mail.gmail.com> <86a32abc0906092113u41381d5ai62212e5d3ef27bc0@mail.gmail.com> <86a32abc0906092143yb0262ddpd737c4fc45b9fa3c@mail.gmail.com> Message-ID: <86a32abc0906092207q3c669c44w38046c6289e41438@mail.gmail.com> Ok I have added a new MySQL example here. http://wiki.freeswitch.org/wiki/Mod_lcr#Sample_Data Diego On Wed, Jun 10, 2009 at 12:43 AM, Diego Viola wrote: > Fixed the issue, I will update the wiki now with a MySQL example. > > Regards, > > Diego > > > On Wed, Jun 10, 2009 at 12:21 AM, Mathieu Rene wrote: > >> If we look at the message again, >> (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) >> REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) >> And put it into words: >> The contrainst names fs_lcr/lcr affecting field carrier_id, and >> referencing the id field of the carriers table, fails. >> In other words, the value you have for carrier_id does not match any >> value of id in the carriers table. >> >> Math >> >> On 10-Jun-09, at 12:13 AM, Diego Viola wrote: >> >> > (`fs_lcr/lcr`, CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) >> > REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/aa3dc0cf/attachment-0002.html From shaheryarkh at googlemail.com Tue Jun 9 22:19:46 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 11:19:46 +0600 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Message-ID: Hi, What kind of problem you are referring to? I am using Skypiax from latest FS trunk revision no. 13613 on CentOS 5.3, Kernel 2.6.18-92.1.22.el5.centos.plusPAE without any problem, the system seems stable and going in production very soon. However, i would like to mention here that i have customized it a bit to add a couple of new commands to allow dynamic Skypiax interface addition and deletion in a running FreeSWITCH process, But instead of changing any existing code i have merely added new code to the exiting, so this shouldn't have resolved the problem you are referring to. The overall performance of both Skypiax and FS are excellent and we are extremely thankful to you guys for developing such great software. If you guys or anyone else need any help in setting up FS or Skypiax on CentOS, do write to me. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli wrote: > Hi all, > > there are problems for mod_skypiax in recent centos, with more than a > handful of concurrent Skype calls. > > Probably the problem is ALSA-related. > > Until it is solved, for production please use Ubuntu 8.04 (see below), > some other Linux distro (and please write here your experience), or > Windows. > > I modified the wiki page to reflect this ( > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > > If someone with CentOS knowledge can chime in I'll be grateful :-). > > Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > all infos, and feel free to contact me directly. > > -giovanni > > > > > > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/cf5ddde8/attachment-0002.html From shaheryarkh at googlemail.com Tue Jun 9 23:33:16 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 12:33:16 +0600 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Message-ID: Sorry, i didn't visited the Jira link you mentioned. Now i know the issue and I have replied it there. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli wrote: > Hi all, > > there are problems for mod_skypiax in recent centos, with more than a > handful of concurrent Skype calls. > > Probably the problem is ALSA-related. > > Until it is solved, for production please use Ubuntu 8.04 (see below), > some other Linux distro (and please write here your experience), or > Windows. > > I modified the wiki page to reflect this ( > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > > If someone with CentOS knowledge can chime in I'll be grateful :-). > > Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > all infos, and feel free to contact me directly. > > -giovanni > > > > > > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/f2512d0d/attachment-0002.html From diego.viola at gmail.com Tue Jun 9 23:38:07 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 02:38:07 -0400 Subject: [Freeswitch-users] mod_nibblebill not set variable nibble_total_billed In-Reply-To: <200906091126.03556.yivzhenko@mksat.net> References: <200906091126.03556.yivzhenko@mksat.net> Message-ID: <86a32abc0906092338hc129269lb7417fc655380c27@mail.gmail.com> Open a Jira. http://jira.freeswitch.org/ On Tue, Jun 9, 2009 at 4:26 AM, Yuriy Ivzhenko wrote: > Some time ago mod_nibblebill was set variable nibble_total_billed after > hangup. > > But after last few updates of module this variable is no more sets. > > Somebody else have this problem? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/11a07e3c/attachment-0002.html From gmaruzz at celliax.org Wed Jun 10 01:37:39 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 10:37:39 +0200 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> Message-ID: <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> Ciao Muhammad! What a good news! Centos is the most stable and performing platform for FS, so I would really love to test and document on the wiki how to have a stable centos mod_skypiax installation. I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE ), and begin to test. In the mean time, do you have any hint, special procedure, etc you have done for having skypiax working well? Please, please, please let be in contact! :-) Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 8:33 AM, Muhammad Shahzad wrote: > Sorry, i didn't visited the Jira link you mentioned. Now i know the issue > and I have replied it there. > > Thank you. > > > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > wrote: >> >> Hi all, >> >> there are problems for mod_skypiax in recent centos, ?with more than a >> handful of concurrent Skype calls. >> >> Probably the problem is ALSA-related. >> >> Until it is solved, for production please use Ubuntu 8.04 (see below), >> some other Linux distro (and please write here your experience), or >> Windows. >> >> I modified the wiki page to reflect this ( >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for >> all infos, and feel free to contact me directly. >> >> -giovanni >> >> >> >> >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 02:16:25 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 15:16:25 +0600 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> Message-ID: Thanks. I didn't make any special arrangements for FS or Skypiax to work on CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE kernel with following commands, root ~# yum update root ~# yum install kernel-PAE i installed PAE kernel just because i wanted to increase System RAM to 8GB before i deploy it for production use, so i can double or even triple Skypiax channels whenever i need so, without system or FS shutdown. As far as a skypiax configuration is concerned, i did modified mod_skypiax.c to add a couple of commands to dynamically add and remove Skypiax interfaces in a running FS process. However, this code does not replaces or changes any previous code. Other then that there is no significant change in configuration steps. Though i did use mod_xml_curl to dynamically update skypiax interface configuration in FS. Thank you. On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad! > > What a good news! > > Centos is the most stable and performing platform for FS, so I would > really love to test and document on the wiki how to have a stable > centos mod_skypiax installation. > > I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE > ), and begin to test. In the mean time, do you have any hint, special > procedure, etc you have done for having skypiax working well? > > Please, please, please let be in contact! :-) > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > Shahzad wrote: > > Sorry, i didn't visited the Jira link you mentioned. Now i know the issue > > and I have replied it there. > > > > Thank you. > > > > > > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Hi all, > >> > >> there are problems for mod_skypiax in recent centos, with more than a > >> handful of concurrent Skype calls. > >> > >> Probably the problem is ALSA-related. > >> > >> Until it is solved, for production please use Ubuntu 8.04 (see below), > >> some other Linux distro (and please write here your experience), or > >> Windows. > >> > >> I modified the wiki page to reflect this ( > >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > >> > >> If someone with CentOS knowledge can chime in I'll be grateful :-). > >> > >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > >> all infos, and feel free to contact me directly. > >> > >> -giovanni > >> > >> > >> > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/a4c49161/attachment-0002.html From gmaruzz at celliax.org Wed Jun 10 02:47:09 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 11:47:09 +0200 Subject: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> Message-ID: <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Ciao Muhammad, first thanks a lot for sharing your experience and help us in making a better software! >From the name of the kernel, seems that you are using centos5.2 is this correct? I just tried centos5.3 (64bit) with centosplus kernel, but no luck. I'm now installing a centos5.2 (64), I will test it with centosplus kernel and with its normal kernel. BTW, I would like *really* a lot to have and integrate your addition to the code (also if it needs some labor from me, no problem). Would you like to send it to me, so I will integrate in the main trunk and you don't have no more to maintain it? (so you can develop other cool features for mod_skypiax ;-) )? -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 11:16 AM, Muhammad Shahzad wrote: > Thanks. I didn't make any special arrangements for FS or Skypiax to work on > CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE > kernel with following commands, > > root ~# yum update > root ~# yum install kernel-PAE > > i installed PAE kernel just because i wanted to increase System RAM to 8GB > before i deploy it for production use, so i can double or even triple > Skypiax channels whenever i need so, without system or FS shutdown. > > As far as a skypiax configuration is concerned, i did modified mod_skypiax.c > to add a couple of commands to dynamically add and remove Skypiax interfaces > in a running FS process. However, this code does not replaces or changes any > previous code. Other then that there is no significant change in > configuration steps. Though i did use mod_xml_curl to dynamically update > skypiax interface configuration in FS. > > > Thank you. > > > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli > wrote: >> >> Ciao Muhammad! >> >> What a good news! >> >> Centos is the most stable and performing platform for FS, so I would >> really love to test and document on the wiki how to have a stable >> centos mod_skypiax installation. >> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >> ), and begin to test. In the mean time, do you have any hint, special >> procedure, etc you have done for having skypiax working well? >> >> Please, please, please let be in contact! :-) >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> Shahzad wrote: >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >> > issue >> > and I have replied it there. >> > >> > Thank you. >> > >> > >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> Hi all, >> >> >> >> there are problems for mod_skypiax in recent centos, ?with more than a >> >> handful of concurrent Skype calls. >> >> >> >> Probably the problem is ALSA-related. >> >> >> >> Until it is solved, for production please use Ubuntu 8.04 (see below), >> >> some other Linux distro (and please write here your experience), or >> >> Windows. >> >> >> >> I modified the wiki page to reflect this ( >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for >> >> all infos, and feel free to contact me directly. >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 03:47:38 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 16:47:38 +0600 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable kernel. I have heard 64bit ALSA drivers have bad sound issues, but never used it personally. As for source code of my modifications, i made those change to develop a customized commercial solution for large European firm, so i would need their permissions to provide you the required official patch. Let me write them an offical request for this. Thank you. On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad, > > first thanks a lot for sharing your experience and help us in making a > better software! > > >From the name of the kernel, seems that you are using centos5.2 is this > correct? > > I just tried centos5.3 (64bit) with centosplus kernel, but no luck. > > I'm now installing a centos5.2 (64), I will test it with centosplus > kernel and with its normal kernel. > > BTW, I would like *really* a lot to have and integrate your addition > to the code (also if it needs some labor from me, no problem). Would > you like to send it to me, so I will integrate in the main trunk and > you don't have no more to maintain it? (so you can develop other cool > features for mod_skypiax ;-) )? > > -giovanni > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 11:16 AM, Muhammad > Shahzad wrote: > > Thanks. I didn't make any special arrangements for FS or Skypiax to work > on > > CentOS 5.3. I only enabled CentOS Plus yum repository and then install > PAE > > kernel with following commands, > > > > root ~# yum update > > root ~# yum install kernel-PAE > > > > i installed PAE kernel just because i wanted to increase System RAM to > 8GB > > before i deploy it for production use, so i can double or even triple > > Skypiax channels whenever i need so, without system or FS shutdown. > > > > As far as a skypiax configuration is concerned, i did modified > mod_skypiax.c > > to add a couple of commands to dynamically add and remove Skypiax > interfaces > > in a running FS process. However, this code does not replaces or changes > any > > previous code. Other then that there is no significant change in > > configuration steps. Though i did use mod_xml_curl to dynamically update > > skypiax interface configuration in FS. > > > > > > Thank you. > > > > > > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Ciao Muhammad! > >> > >> What a good news! > >> > >> Centos is the most stable and performing platform for FS, so I would > >> really love to test and document on the wiki how to have a stable > >> centos mod_skypiax installation. > >> > >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE > >> ), and begin to test. In the mean time, do you have any hint, special > >> procedure, etc you have done for having skypiax working well? > >> > >> Please, please, please let be in contact! :-) > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> > >> > >> > >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > >> Shahzad wrote: > >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the > >> > issue > >> > and I have replied it there. > >> > > >> > Thank you. > >> > > >> > > >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > >> > > >> > wrote: > >> >> > >> >> Hi all, > >> >> > >> >> there are problems for mod_skypiax in recent centos, with more than > a > >> >> handful of concurrent Skype calls. > >> >> > >> >> Probably the problem is ALSA-related. > >> >> > >> >> Until it is solved, for production please use Ubuntu 8.04 (see > below), > >> >> some other Linux distro (and please write here your experience), or > >> >> Windows. > >> >> > >> >> I modified the wiki page to reflect this ( > >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) > >> >> > >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). > >> >> > >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for > >> >> all infos, and feel free to contact me directly. > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> ========================================= > >> >> www.celliax.org > >> >> via Pierlombardo 9, 20135 Milano > >> >> Italy > >> >> gmaruzz at celliax dot org > >> >> Cell : +39-347-2665618 > >> >> Fax : +39-02-87390039 > >> >> > >> >> _______________________________________________ > >> >> Freeswitch-users mailing list > >> >> Freeswitch-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Muhammad Shahzad > >> > ----------------------------------- > >> > CISCO Rich Media Communication Specialist (CRMCS) > >> > CISCO Certified Network Associate (CCNA) > >> > Cell: +92 334 422 40 88 > >> > MSN: shari_786pk at hotmail.com > >> > Email: shaheryarkh at googlemail.com > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/e507a830/attachment-0002.html From gmaruzz at celliax.org Wed Jun 10 05:27:46 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 14:27:46 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: <7b197bef0906100527w4876413bif292a7dcadf60fd4@mail.gmail.com> Thanks a lot Muhammad, and please let that firm know the advantages of having customizations included into mainstream ;-). OK, I will try 32 bit too, and see if there is differences. So, you started fom a fresh install of centos5.2, then you installed the PAE kernel. Is this right? Pay attention, because if you do an "yum update" now, it will install the "128" kernel, no more the "92", and maybe this will break something..... Anyway, I'm investigating, and please let me know if you'll have additional infos. Hope to hear from you soon, -giovanni On Wed, Jun 10, 2009 at 12:47 PM, Muhammad Shahzad wrote: > I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable > kernel. I have heard 64bit ALSA drivers have bad sound issues, but never > used it personally. > > As for source code of my modifications, i made those change to develop a > customized commercial solution for large European firm, so i would need > their permissions to provide you the required official patch. Let me write > them an offical request for this. > > Thank you. > > > On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli > wrote: >> >> Ciao Muhammad, >> >> first thanks a lot for sharing your experience and help us in making a >> better software! >> >> >From the name of the kernel, seems that you are using centos5.2 is this >> correct? >> >> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >> >> I'm now installing a centos5.2 (64), I will test it with centosplus >> kernel and with its normal kernel. >> >> BTW, I would like *really* a lot to have and integrate your addition >> to the code (also if it needs some labor from me, no problem). Would >> you like to send it to me, so I will integrate in the main trunk and >> you don't have no more to maintain it? (so you can develop other cool >> features for mod_skypiax ;-) )? >> >> -giovanni >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >> Shahzad wrote: >> > Thanks. I didn't make any special arrangements for FS or Skypiax to work >> > on >> > CentOS 5.3. I only enabled CentOS Plus yum repository and then install >> > PAE >> > kernel with following commands, >> > >> > root ~# yum update >> > root ~# yum install kernel-PAE >> > >> > i installed PAE kernel just because i wanted to increase System RAM to >> > 8GB >> > before i deploy it for production use, so i can double or even triple >> > Skypiax channels whenever i need so, without system or FS shutdown. >> > >> > As far as a skypiax configuration is concerned, i did modified >> > mod_skypiax.c >> > to add a couple of commands to dynamically add and remove Skypiax >> > interfaces >> > in a running FS process. However, this code does not replaces or changes >> > any >> > previous code. Other then that there is no significant change in >> > configuration steps. Though i did use mod_xml_curl to dynamically update >> > skypiax interface configuration in FS. >> > >> > >> > Thank you. >> > >> > >> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> Ciao Muhammad! >> >> >> >> What a good news! >> >> >> >> Centos is the most stable and performing platform for FS, so I would >> >> really love to test and document on the wiki how to have a stable >> >> centos mod_skypiax installation. >> >> >> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >> >> ), and begin to test. In the mean time, do you have any hint, special >> >> procedure, etc you have done for having skypiax working well? >> >> >> >> Please, please, please let be in contact! :-) >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> >> >> >> >> >> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> >> Shahzad wrote: >> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >> >> > issue >> >> > and I have replied it there. >> >> > >> >> > Thank you. >> >> > >> >> > >> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> >> > >> >> > wrote: >> >> >> >> >> >> Hi all, >> >> >> >> >> >> there are problems for mod_skypiax in recent centos, ?with more than >> >> >> a >> >> >> handful of concurrent Skype calls. >> >> >> >> >> >> Probably the problem is ALSA-related. >> >> >> >> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >> >> >> below), >> >> >> some other Linux distro (and please write here your experience), or >> >> >> Windows. >> >> >> >> >> >> I modified the wiki page to reflect this ( >> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> >> >> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> >> >> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for >> >> >> all infos, and feel free to contact me directly. >> >> >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> >> >> Giovanni Maruzzelli >> >> >> ========================================= >> >> >> www.celliax.org >> >> >> via Pierlombardo 9, 20135 Milano >> >> >> Italy >> >> >> gmaruzz at celliax dot org >> >> >> Cell : +39-347-2665618 >> >> >> Fax : +39-02-87390039 >> >> >> >> >> >> _______________________________________________ >> >> >> Freeswitch-users mailing list >> >> >> Freeswitch-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Muhammad Shahzad >> >> > ----------------------------------- >> >> > CISCO Rich Media Communication Specialist (CRMCS) >> >> > CISCO Certified Network Associate (CCNA) >> >> > Cell: +92 334 422 40 88 >> >> > MSN: shari_786pk at hotmail.com >> >> > Email: shaheryarkh at googlemail.com >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 05:29:44 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 18:29:44 +0600 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: I am glad to share the patch to enable dynamic Skypiax interfaces in FS. Please do note that however, that i started working on it on May 22, 2009. So any officaily changes made to mod_skypiax.c since then will not appear in it and will be lost if you apply this patch blindly. I request Giovanni Maruzzelli to carefully merge this patch in main stream code before committing it to FS SVN. Thank you. On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable > kernel. I have heard 64bit ALSA drivers have bad sound issues, but never > used it personally. > > As for source code of my modifications, i made those change to develop a > customized commercial solution for large European firm, so i would need > their permissions to provide you the required official patch. Let me write > them an offical request for this. > > Thank you. > > > > On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli wrote: > >> Ciao Muhammad, >> >> first thanks a lot for sharing your experience and help us in making a >> better software! >> >> >From the name of the kernel, seems that you are using centos5.2 is this >> correct? >> >> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >> >> I'm now installing a centos5.2 (64), I will test it with centosplus >> kernel and with its normal kernel. >> >> BTW, I would like *really* a lot to have and integrate your addition >> to the code (also if it needs some labor from me, no problem). Would >> you like to send it to me, so I will integrate in the main trunk and >> you don't have no more to maintain it? (so you can develop other cool >> features for mod_skypiax ;-) )? >> >> -giovanni >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >> Shahzad wrote: >> > Thanks. I didn't make any special arrangements for FS or Skypiax to work >> on >> > CentOS 5.3. I only enabled CentOS Plus yum repository and then install >> PAE >> > kernel with following commands, >> > >> > root ~# yum update >> > root ~# yum install kernel-PAE >> > >> > i installed PAE kernel just because i wanted to increase System RAM to >> 8GB >> > before i deploy it for production use, so i can double or even triple >> > Skypiax channels whenever i need so, without system or FS shutdown. >> > >> > As far as a skypiax configuration is concerned, i did modified >> mod_skypiax.c >> > to add a couple of commands to dynamically add and remove Skypiax >> interfaces >> > in a running FS process. However, this code does not replaces or changes >> any >> > previous code. Other then that there is no significant change in >> > configuration steps. Though i did use mod_xml_curl to dynamically update >> > skypiax interface configuration in FS. >> > >> > >> > Thank you. >> > >> > >> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli < >> gmaruzz at celliax.org> >> > wrote: >> >> >> >> Ciao Muhammad! >> >> >> >> What a good news! >> >> >> >> Centos is the most stable and performing platform for FS, so I would >> >> really love to test and document on the wiki how to have a stable >> >> centos mod_skypiax installation. >> >> >> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >> >> ), and begin to test. In the mean time, do you have any hint, special >> >> procedure, etc you have done for having skypiax working well? >> >> >> >> Please, please, please let be in contact! :-) >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> >> >> >> >> >> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> >> Shahzad wrote: >> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >> >> > issue >> >> > and I have replied it there. >> >> > >> >> > Thank you. >> >> > >> >> > >> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> >> > >> >> > wrote: >> >> >> >> >> >> Hi all, >> >> >> >> >> >> there are problems for mod_skypiax in recent centos, with more than >> a >> >> >> handful of concurrent Skype calls. >> >> >> >> >> >> Probably the problem is ALSA-related. >> >> >> >> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >> below), >> >> >> some other Linux distro (and please write here your experience), or >> >> >> Windows. >> >> >> >> >> >> I modified the wiki page to reflect this ( >> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >> >> >> >> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >> >> >> >> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34for >> >> >> all infos, and feel free to contact me directly. >> >> >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> >> >> Giovanni Maruzzelli >> >> >> ========================================= >> >> >> www.celliax.org >> >> >> via Pierlombardo 9, 20135 Milano >> >> >> Italy >> >> >> gmaruzz at celliax dot org >> >> >> Cell : +39-347-2665618 >> >> >> Fax : +39-02-87390039 >> >> >> >> >> >> _______________________________________________ >> >> >> Freeswitch-users mailing list >> >> >> Freeswitch-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Muhammad Shahzad >> >> > ----------------------------------- >> >> > CISCO Rich Media Communication Specialist (CRMCS) >> >> > CISCO Certified Network Associate (CCNA) >> >> > Cell: +92 334 422 40 88 >> >> > MSN: shari_786pk at hotmail.com >> >> > Email: shaheryarkh at googlemail.com >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/efce2963/attachment-0002.html From gmaruzz at celliax.org Wed Jun 10 05:33:34 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 14:33:34 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> Ciao Muhammad, you're faster than light :-)! the patch will be integrated very soon, I'll let you know when I'm done it. Keep enhancements, patches, bug fixes, etc flowing! thanks again, and thanks to the firm that so quickly understood and authorized you, -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 2:29 PM, Muhammad Shahzad wrote: > I am glad to share the patch to enable dynamic Skypiax interfaces in FS. > Please do note that however, that i started working on it on May 22, 2009. > So any officaily changes made to mod_skypiax.c since then will not appear in > it and will be lost if you apply this patch blindly. > > I request Giovanni Maruzzelli to carefully merge this patch in main stream > code before committing it to FS SVN. > > Thank you. > > > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad > wrote: >> >> I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable >> kernel. I have heard 64bit ALSA drivers have bad sound issues, but never >> used it personally. >> >> As for source code of my modifications, i made those change to develop a >> customized commercial solution for large European firm, so i would need >> their permissions to provide you the required official patch. Let me write >> them an offical request for this. >> >> Thank you. >> >> >> On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli >> wrote: >>> >>> Ciao Muhammad, >>> >>> first thanks a lot for sharing your experience and help us in making a >>> better software! >>> >>> >From the name of the kernel, seems that you are using centos5.2 is this >>> correct? >>> >>> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >>> >>> I'm now installing a centos5.2 (64), I will test it with centosplus >>> kernel and with its normal kernel. >>> >>> BTW, I would like *really* a lot to have and integrate your addition >>> to the code (also if it needs some labor from me, no problem). Would >>> you like to send it to me, so I will integrate in the main trunk and >>> you don't have no more to maintain it? (so you can develop other cool >>> features for mod_skypiax ;-) )? >>> >>> -giovanni >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> ========================================= >>> www.celliax.org >>> via Pierlombardo 9, 20135 Milano >>> Italy >>> gmaruzz at celliax dot org >>> Cell : +39-347-2665618 >>> Fax : +39-02-87390039 >>> >>> >>> >>> >>> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >>> Shahzad wrote: >>> > Thanks. I didn't make any special arrangements for FS or Skypiax to >>> > work on >>> > CentOS 5.3. I only enabled CentOS Plus yum repository and then install >>> > PAE >>> > kernel with following commands, >>> > >>> > root ~# yum update >>> > root ~# yum install kernel-PAE >>> > >>> > i installed PAE kernel just because i wanted to increase System RAM to >>> > 8GB >>> > before i deploy it for production use, so i can double or even triple >>> > Skypiax channels whenever i need so, without system or FS shutdown. >>> > >>> > As far as a skypiax configuration is concerned, i did modified >>> > mod_skypiax.c >>> > to add a couple of commands to dynamically add and remove Skypiax >>> > interfaces >>> > in a running FS process. However, this code does not replaces or >>> > changes any >>> > previous code. Other then that there is no significant change in >>> > configuration steps. Though i did use mod_xml_curl to dynamically >>> > update >>> > skypiax interface configuration in FS. >>> > >>> > >>> > Thank you. >>> > >>> > >>> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli >>> > >>> > wrote: >>> >> >>> >> Ciao Muhammad! >>> >> >>> >> What a good news! >>> >> >>> >> Centos is the most stable and performing platform for FS, so I would >>> >> really love to test and document on the wiki how to have a stable >>> >> centos mod_skypiax installation. >>> >> >>> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE >>> >> ), and begin to test. In the mean time, do you have any hint, special >>> >> procedure, etc you have done for having skypiax working well? >>> >> >>> >> Please, please, please let be in contact! :-) >>> >> >>> >> >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> ========================================= >>> >> www.celliax.org >>> >> via Pierlombardo 9, 20135 Milano >>> >> Italy >>> >> gmaruzz at celliax dot org >>> >> Cell : +39-347-2665618 >>> >> Fax : +39-02-87390039 >>> >> >>> >> >>> >> >>> >> >>> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >>> >> Shahzad wrote: >>> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know the >>> >> > issue >>> >> > and I have replied it there. >>> >> > >>> >> > Thank you. >>> >> > >>> >> > >>> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >>> >> > >>> >> > wrote: >>> >> >> >>> >> >> Hi all, >>> >> >> >>> >> >> there are problems for mod_skypiax in recent centos, ?with more >>> >> >> than a >>> >> >> handful of concurrent Skype calls. >>> >> >> >>> >> >> Probably the problem is ALSA-related. >>> >> >> >>> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >>> >> >> below), >>> >> >> some other Linux distro (and please write here your experience), or >>> >> >> Windows. >>> >> >> >>> >> >> I modified the wiki page to reflect this ( >>> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) >>> >> >> >>> >> >> If someone with CentOS knowledge can chime in I'll be grateful :-). >>> >> >> >>> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 >>> >> >> for >>> >> >> all infos, and feel free to contact me directly. >>> >> >> >>> >> >> -giovanni >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> Sincerely, >>> >> >> >>> >> >> Giovanni Maruzzelli >>> >> >> ========================================= >>> >> >> www.celliax.org >>> >> >> via Pierlombardo 9, 20135 Milano >>> >> >> Italy >>> >> >> gmaruzz at celliax dot org >>> >> >> Cell : +39-347-2665618 >>> >> >> Fax : +39-02-87390039 >>> >> >> >>> >> >> _______________________________________________ >>> >> >> Freeswitch-users mailing list >>> >> >> Freeswitch-users at lists.freeswitch.org >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> >>> >> >> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> http://www.freeswitch.org >>> >> > >>> >> > >>> >> > >>> >> > -- >>> >> > Muhammad Shahzad >>> >> > ----------------------------------- >>> >> > CISCO Rich Media Communication Specialist (CRMCS) >>> >> > CISCO Certified Network Associate (CCNA) >>> >> > Cell: +92 334 422 40 88 >>> >> > MSN: shari_786pk at hotmail.com >>> >> > Email: shaheryarkh at googlemail.com >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > -- >>> > Muhammad Shahzad >>> > ----------------------------------- >>> > CISCO Rich Media Communication Specialist (CRMCS) >>> > CISCO Certified Network Associate (CCNA) >>> > Cell: +92 334 422 40 88 >>> > MSN: shari_786pk at hotmail.com >>> > Email: shaheryarkh at googlemail.com >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> _______________________________________________ >>> Freeswitch-dev mailing list >>> Freeswitch-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Wed Jun 10 06:01:57 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 10 Jun 2009 19:01:57 +0600 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> Message-ID: You are welcome. Let me elaborate my setup here, I have two machines, one for development, this is basically my lenovo 3000 N200 laptop, it has following specs, 1. Intel 1.6 GHz with 1GB RAM. 2. CentOS 5.3 with Kernel 2.6.18-128.1.6.el5. 3. FS SVN revision Revision ID 13613. root ~# uname -a Linux localhost.localdomain 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:19:18 EDT 2009 i686 i686 i386 GNU/Linux root ~# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m I am using this machine extensively for my development projects, including Skypiax. Yesterday i gave a presentation to the board of directors of the said firm, regarding existing status of my project. They tested the setup with 2-3 concurrent SIP -> SKYPIAX and then SKYPIAX to SIP calls without any problem. So, i believe this configuration works without any sound issue...! The second machine is my test machine in a remote data center. I didn't prepare this machine, however, from SSH console i can see it has following specs, 1. Intel(R) Xeon(R) CPU E5405 @ 2.00GHz with 4 GB of RAM. 2. CentOS 5.3 with kernel 2.6.18-92.1.22.el5.centos.plusPAE. 3. FS SVN revision Revision ID 13613. root ~# uname -a Linux localhost.localdomain 2.6.18-92.1.22.el5.centos.plusPAE #1 SMP Wed Dec 17 11:32:56 EST 2008 i686 i686 i386 GNU/Linux root ~# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m Each machine that i use always, get update with yum update command BEFORE i do anything else on it. Hope this info will be helpful for you. Can you give me step by step procedure of your testing that is producing this bad sound result? I would like to perform this test on my both machines and see if i get the same results too. Thank you. On Wed, Jun 10, 2009 at 6:33 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad, > you're faster than light :-)! > > the patch will be integrated very soon, I'll let you know when I'm done it. > > Keep enhancements, patches, bug fixes, etc flowing! > > thanks again, and thanks to the firm that so quickly understood and > authorized you, > > -giovanni > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 2:29 PM, Muhammad > Shahzad wrote: > > I am glad to share the patch to enable dynamic Skypiax interfaces in FS. > > Please do note that however, that i started working on it on May 22, > 2009. > > So any officaily changes made to mod_skypiax.c since then will not appear > in > > it and will be lost if you apply this patch blindly. > > > > I request Giovanni Maruzzelli to carefully merge this patch in main > stream > > code before committing it to FS SVN. > > > > Thank you. > > > > > > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad > > wrote: > >> > >> I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable > >> kernel. I have heard 64bit ALSA drivers have bad sound issues, but never > >> used it personally. > >> > >> As for source code of my modifications, i made those change to develop a > >> customized commercial solution for large European firm, so i would need > >> their permissions to provide you the required official patch. Let me > write > >> them an offical request for this. > >> > >> Thank you. > >> > >> > >> On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > >> wrote: > >>> > >>> Ciao Muhammad, > >>> > >>> first thanks a lot for sharing your experience and help us in making a > >>> better software! > >>> > >>> >From the name of the kernel, seems that you are using centos5.2 is > this > >>> correct? > >>> > >>> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. > >>> > >>> I'm now installing a centos5.2 (64), I will test it with centosplus > >>> kernel and with its normal kernel. > >>> > >>> BTW, I would like *really* a lot to have and integrate your addition > >>> to the code (also if it needs some labor from me, no problem). Would > >>> you like to send it to me, so I will integrate in the main trunk and > >>> you don't have no more to maintain it? (so you can develop other cool > >>> features for mod_skypiax ;-) )? > >>> > >>> -giovanni > >>> > >>> Sincerely, > >>> > >>> Giovanni Maruzzelli > >>> ========================================= > >>> www.celliax.org > >>> via Pierlombardo 9, 20135 Milano > >>> Italy > >>> gmaruzz at celliax dot org > >>> Cell : +39-347-2665618 > >>> Fax : +39-02-87390039 > >>> > >>> > >>> > >>> > >>> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad > >>> Shahzad wrote: > >>> > Thanks. I didn't make any special arrangements for FS or Skypiax to > >>> > work on > >>> > CentOS 5.3. I only enabled CentOS Plus yum repository and then > install > >>> > PAE > >>> > kernel with following commands, > >>> > > >>> > root ~# yum update > >>> > root ~# yum install kernel-PAE > >>> > > >>> > i installed PAE kernel just because i wanted to increase System RAM > to > >>> > 8GB > >>> > before i deploy it for production use, so i can double or even triple > >>> > Skypiax channels whenever i need so, without system or FS shutdown. > >>> > > >>> > As far as a skypiax configuration is concerned, i did modified > >>> > mod_skypiax.c > >>> > to add a couple of commands to dynamically add and remove Skypiax > >>> > interfaces > >>> > in a running FS process. However, this code does not replaces or > >>> > changes any > >>> > previous code. Other then that there is no significant change in > >>> > configuration steps. Though i did use mod_xml_curl to dynamically > >>> > update > >>> > skypiax interface configuration in FS. > >>> > > >>> > > >>> > Thank you. > >>> > > >>> > > >>> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli > >>> > > >>> > wrote: > >>> >> > >>> >> Ciao Muhammad! > >>> >> > >>> >> What a good news! > >>> >> > >>> >> Centos is the most stable and performing platform for FS, so I would > >>> >> really love to test and document on the wiki how to have a stable > >>> >> centos mod_skypiax installation. > >>> >> > >>> >> I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE > >>> >> ), and begin to test. In the mean time, do you have any hint, > special > >>> >> procedure, etc you have done for having skypiax working well? > >>> >> > >>> >> Please, please, please let be in contact! :-) > >>> >> > >>> >> > >>> >> Sincerely, > >>> >> > >>> >> Giovanni Maruzzelli > >>> >> ========================================= > >>> >> www.celliax.org > >>> >> via Pierlombardo 9, 20135 Milano > >>> >> Italy > >>> >> gmaruzz at celliax dot org > >>> >> Cell : +39-347-2665618 > >>> >> Fax : +39-02-87390039 > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > >>> >> Shahzad wrote: > >>> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know > the > >>> >> > issue > >>> >> > and I have replied it there. > >>> >> > > >>> >> > Thank you. > >>> >> > > >>> >> > > >>> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > >>> >> > > >>> >> > wrote: > >>> >> >> > >>> >> >> Hi all, > >>> >> >> > >>> >> >> there are problems for mod_skypiax in recent centos, with more > >>> >> >> than a > >>> >> >> handful of concurrent Skype calls. > >>> >> >> > >>> >> >> Probably the problem is ALSA-related. > >>> >> >> > >>> >> >> Until it is solved, for production please use Ubuntu 8.04 (see > >>> >> >> below), > >>> >> >> some other Linux distro (and please write here your experience), > or > >>> >> >> Windows. > >>> >> >> > >>> >> >> I modified the wiki page to reflect this ( > >>> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk) > >>> >> >> > >>> >> >> If someone with CentOS knowledge can chime in I'll be grateful > :-). > >>> >> >> > >>> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 > >>> >> >> for > >>> >> >> all infos, and feel free to contact me directly. > >>> >> >> > >>> >> >> -giovanni > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> > >>> >> >> Sincerely, > >>> >> >> > >>> >> >> Giovanni Maruzzelli > >>> >> >> ========================================= > >>> >> >> www.celliax.org > >>> >> >> via Pierlombardo 9, 20135 Milano > >>> >> >> Italy > >>> >> >> gmaruzz at celliax dot org > >>> >> >> Cell : +39-347-2665618 > >>> >> >> Fax : +39-02-87390039 > >>> >> >> > >>> >> >> _______________________________________________ > >>> >> >> Freeswitch-users mailing list > >>> >> >> Freeswitch-users at lists.freeswitch.org > >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> >> > >>> >> >> > >>> >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> >> http://www.freeswitch.org > >>> >> > > >>> >> > > >>> >> > > >>> >> > -- > >>> >> > Muhammad Shahzad > >>> >> > ----------------------------------- > >>> >> > CISCO Rich Media Communication Specialist (CRMCS) > >>> >> > CISCO Certified Network Associate (CCNA) > >>> >> > Cell: +92 334 422 40 88 > >>> >> > MSN: shari_786pk at hotmail.com > >>> >> > Email: shaheryarkh at googlemail.com > >>> >> > > >>> >> > _______________________________________________ > >>> >> > Freeswitch-users mailing list > >>> >> > Freeswitch-users at lists.freeswitch.org > >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > > >>> >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> > http://www.freeswitch.org > >>> >> > > >>> >> > > >>> >> > >>> >> _______________________________________________ > >>> >> Freeswitch-users mailing list > >>> >> Freeswitch-users at lists.freeswitch.org > >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> http://www.freeswitch.org > >>> > > >>> > > >>> > > >>> > -- > >>> > Muhammad Shahzad > >>> > ----------------------------------- > >>> > CISCO Rich Media Communication Specialist (CRMCS) > >>> > CISCO Certified Network Associate (CCNA) > >>> > Cell: +92 334 422 40 88 > >>> > MSN: shari_786pk at hotmail.com > >>> > Email: shaheryarkh at googlemail.com > >>> > > >>> > _______________________________________________ > >>> > Freeswitch-users mailing list > >>> > Freeswitch-users at lists.freeswitch.org > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > > >>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > > >>> > > >>> > >>> _______________________________________________ > >>> Freeswitch-dev mailing list > >>> Freeswitch-dev at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > >>> http://www.freeswitch.org > >> > >> > >> > >> -- > >> Muhammad Shahzad > >> ----------------------------------- > >> CISCO Rich Media Communication Specialist (CRMCS) > >> CISCO Certified Network Associate (CCNA) > >> Cell: +92 334 422 40 88 > >> MSN: shari_786pk at hotmail.com > >> Email: shaheryarkh at googlemail.com > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/8ff45edc/attachment-0002.html From teqx at yahoo.com Wed Jun 10 06:05:03 2009 From: teqx at yahoo.com (zigurds) Date: Wed, 10 Jun 2009 06:05:03 -0700 (PDT) Subject: [Freeswitch-users] how to determine who is hangup the call Message-ID: <23961888.post@talk.nabble.com> Hi, How to determine from channel events, that are sent through event_socket, what party have terminated the call? If I call for example from 103 to 104 and in first time 104 hang up, but in second time 103 hang up, in both cases CHANNEL_HANGUP is sent at first to 104: Event-Name: CHANNEL_HANGUP Core-UUID: 8729f181-3325-4373-8030-537bc090e9d3 ... Event-Date-Timestamp: 1244634602734687 Event-Calling-File: switch_core_state_machine.c Event-Calling-Function: switch_core_session_run Event-Calling-Line-Number: 469 Hangup-Cause: NORMAL_CLEARING Channel-State: CS_HANGUP Channel-State-Number: 10 Channel-Name: sofia/Test/104 Unique-ID: 17cdd5b7-8949-44d4-bbb9-5d17a713811c Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered ... Thanks, Zigurds -- View this message in context: http://www.nabble.com/how-to-determine-who-is-hangup-the-call-tp23961888p23961888.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Wed Jun 10 06:10:50 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 21:10:50 +0800 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: Glad to see the patch, I have been waiting for a long time :) btw, I have changed the start scripts a bit to start X and skype separately, glad to share it if someone interested. On Jun 10, 2009, at 8:29 PM, Muhammad Shahzad wrote: > I am glad to share the patch to enable dynamic Skypiax interfaces in > FS. Please do note that however, that i started working on it on May > 22, 2009. So any officaily changes made to mod_skypiax.c since then > will not appear in it and will be lost if you apply this patch > blindly. > > I request Giovanni Maruzzelli to carefully merge this patch in main > stream code before committing it to FS SVN. > > Thank you. > > > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad > wrote: > I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE > enable kernel. I have heard 64bit ALSA drivers have bad sound > issues, but never used it personally. > > As for source code of my modifications, i made those change to > develop a customized commercial solution for large European firm, so > i would need their permissions to provide you the required official > patch. Let me write them an offical request for this. > > Thank you. > > > > On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli > wrote: > Ciao Muhammad, > > first thanks a lot for sharing your experience and help us in making a > better software! > > >From the name of the kernel, seems that you are using centos5.2 is > this correct? > > I just tried centos5.3 (64bit) with centosplus kernel, but no luck. > > I'm now installing a centos5.2 (64), I will test it with centosplus > kernel and with its normal kernel. > > BTW, I would like *really* a lot to have and integrate your addition > to the code (also if it needs some labor from me, no problem). Would > you like to send it to me, so I will integrate in the main trunk and > you don't have no more to maintain it? (so you can develop other cool > features for mod_skypiax ;-) )? > > -giovanni > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Wed, Jun 10, 2009 at 11:16 AM, Muhammad > Shahzad wrote: > > Thanks. I didn't make any special arrangements for FS or Skypiax > to work on > > CentOS 5.3. I only enabled CentOS Plus yum repository and then > install PAE > > kernel with following commands, > > > > root ~# yum update > > root ~# yum install kernel-PAE > > > > i installed PAE kernel just because i wanted to increase System > RAM to 8GB > > before i deploy it for production use, so i can double or even > triple > > Skypiax channels whenever i need so, without system or FS shutdown. > > > > As far as a skypiax configuration is concerned, i did modified > mod_skypiax.c > > to add a couple of commands to dynamically add and remove Skypiax > interfaces > > in a running FS process. However, this code does not replaces or > changes any > > previous code. Other then that there is no significant change in > > configuration steps. Though i did use mod_xml_curl to dynamically > update > > skypiax interface configuration in FS. > > > > > > Thank you. > > > > > > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli > > > wrote: > >> > >> Ciao Muhammad! > >> > >> What a good news! > >> > >> Centos is the most stable and performing platform for FS, so I > would > >> really love to test and document on the wiki how to have a stable > >> centos mod_skypiax installation. > >> > >> I'll find out your kernel ( Kernel > 2.6.18-92.1.22.el5.centos.plusPAE > >> ), and begin to test. In the mean time, do you have any hint, > special > >> procedure, etc you have done for having skypiax working well? > >> > >> Please, please, please let be in contact! :-) > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> > >> > >> > >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad > >> Shahzad wrote: > >> > Sorry, i didn't visited the Jira link you mentioned. Now i know > the > >> > issue > >> > and I have replied it there. > >> > > >> > Thank you. > >> > > >> > > >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli > >> > > >> > wrote: > >> >> > >> >> Hi all, > >> >> > >> >> there are problems for mod_skypiax in recent centos, with > more than a > >> >> handful of concurrent Skype calls. > >> >> > >> >> Probably the problem is ALSA-related. > >> >> > >> >> Until it is solved, for production please use Ubuntu 8.04 (see > below), > >> >> some other Linux distro (and please write here your > experience), or > >> >> Windows. > >> >> > >> >> I modified the wiki page to reflect this ( > >> >> http://wiki.freeswitch.org/wiki/ > Skypiax_Skype_Endpoint_and_Trunk ) > >> >> > >> >> If someone with CentOS knowledge can chime in I'll be > grateful :-). > >> >> > >> >> Please see Jira: http://jira.freeswitch.org/browse/ > MODSKYPIAX-34 for > >> >> all infos, and feel free to contact me directly. > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> ========================================= > >> >> www.celliax.org > >> >> via Pierlombardo 9, 20135 Milano > >> >> Italy > >> >> gmaruzz at celliax dot org > >> >> Cell : +39-347-2665618 > >> >> Fax : +39-02-87390039 > >> >> > >> >> _______________________________________________ > >> >> Freeswitch-users mailing list > >> >> Freeswitch-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Muhammad Shahzad > >> > ----------------------------------- > >> > CISCO Rich Media Communication Specialist (CRMCS) > >> > CISCO Certified Network Associate (CCNA) > >> > Cell: +92 334 422 40 88 > >> > MSN: shari_786pk at hotmail.com > >> > Email: shaheryarkh at googlemail.com > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/6419b0f6/attachment-0002.html From gmaruzz at celliax.org Wed Jun 10 06:28:23 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 15:28:23 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> Message-ID: <7b197bef0906100628y25948a97x59af0ebd02c47d71@mail.gmail.com> On Wed, Jun 10, 2009 at 3:10 PM, dujinfang wrote: > btw, I have changed the start scripts a bit to start X and skype?separately, > glad to share it if someone?interested. I'm interested! :-) From gmaruzz at celliax.org Wed Jun 10 06:37:03 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 10 Jun 2009 15:37:03 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> <7b197bef0906100533y5f90c7fanff2e2dd29738fc07@mail.gmail.com> Message-ID: <7b197bef0906100637q7068ea56h666048867d968721@mail.gmail.com> On Wed, Jun 10, 2009 at 3:01 PM, Muhammad Shahzad wrote: > I am using this machine extensively for my development projects, including > Skypiax. Yesterday i gave a presentation to the board of directors of the > said firm, regarding existing status of my project. They tested the setup > with 2-3 concurrent SIP -> SKYPIAX and then SKYPIAX to SIP calls without any > problem. So, i believe this configuration works without any sound issue...! Until some times ago, I was used to test with 32 bit centos, and I've not seen problems (I tested with centos5.2). Now I'm installing 5.3 32 bit, and I'll report on my results On 64bit, on both centos5.2 and 5.3 I'm seeing problems with more than 6-10 concurrent calls, with less calls it was ok. BTW, it can be it is a problem with my hard disk (that is seen as an ide instead than as a SATA). I will check this too. I'll report back soon on the various issues... > > The second machine is my test machine in a remote data center. I didn't > prepare this machine, however, from SSH console i can see it has following > specs, > > 1. Intel(R) Xeon(R) CPU E5405? @ 2.00GHz with 4 GB of RAM. > 2. CentOS 5.3 with kernel 2.6.18-92.1.22.el5.centos.plusPAE. > 3. FS SVN revision Revision ID 13613. > > root ~# uname -a > Linux localhost.localdomain 2.6.18-92.1.22.el5.centos.plusPAE #1 SMP Wed Dec > 17 11:32:56 EST 2008 i686 i686 i386 GNU/Linux > > root ~# cat /etc/issue > CentOS release 5.3 (Final) > Kernel \r on an \m > > > Each machine that i use always, get update with yum update command BEFORE i > do anything else on it. > > Hope this info will be helpful for you. > > Can you give me step by step procedure of your testing that is producing > this bad sound result? I would like to perform this test on my both machines > and see if i get the same results too. > > Thank you. > > > On Wed, Jun 10, 2009 at 6:33 PM, Giovanni Maruzzelli > wrote: >> >> Ciao Muhammad, >> you're faster than light :-)! >> >> the patch will be integrated very soon, I'll let you know when I'm done >> it. >> >> Keep enhancements, patches, bug fixes, etc flowing! >> >> thanks again, and thanks to the firm that so quickly understood and >> authorized you, >> >> -giovanni >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Wed, Jun 10, 2009 at 2:29 PM, Muhammad >> Shahzad wrote: >> > I am glad to share the patch to enable dynamic Skypiax interfaces in FS. >> > Please do note that however, that i started working on it on May 22, >> > 2009. >> > So any officaily changes made to mod_skypiax.c since then will not >> > appear in >> > it and will be lost if you apply this patch blindly. >> > >> > I request Giovanni Maruzzelli to carefully merge this patch in main >> > stream >> > code before committing it to FS SVN. >> > >> > Thank you. >> > >> > >> > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad >> > wrote: >> >> >> >> I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE >> >> enable >> >> kernel. I have heard 64bit ALSA drivers have bad sound issues, but >> >> never >> >> used it personally. >> >> >> >> As for source code of my modifications, i made those change to develop >> >> a >> >> customized commercial solution for large European firm, so i would need >> >> their permissions to provide you the required official patch. Let me >> >> write >> >> them an offical request for this. >> >> >> >> Thank you. >> >> >> >> >> >> On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli >> >> >> >> wrote: >> >>> >> >>> Ciao Muhammad, >> >>> >> >>> first thanks a lot for sharing your experience and help us in making a >> >>> better software! >> >>> >> >>> >From the name of the kernel, seems that you are using centos5.2 is >> >>> this >> >>> correct? >> >>> >> >>> I just tried centos5.3 (64bit) with centosplus kernel, but no luck. >> >>> >> >>> I'm now installing a centos5.2 (64), I will test it with centosplus >> >>> kernel and with its normal kernel. >> >>> >> >>> BTW, I would like *really* a lot to have and integrate your addition >> >>> to the code (also if it needs some labor from me, no problem). Would >> >>> you like to send it to me, so I will integrate in the main trunk and >> >>> you don't have no more to maintain it? (so you can develop other cool >> >>> features for mod_skypiax ;-) )? >> >>> >> >>> -giovanni >> >>> >> >>> Sincerely, >> >>> >> >>> Giovanni Maruzzelli >> >>> ========================================= >> >>> www.celliax.org >> >>> via Pierlombardo 9, 20135 Milano >> >>> Italy >> >>> gmaruzz at celliax dot org >> >>> Cell : +39-347-2665618 >> >>> Fax : +39-02-87390039 >> >>> >> >>> >> >>> >> >>> >> >>> On Wed, Jun 10, 2009 at 11:16 AM, Muhammad >> >>> Shahzad wrote: >> >>> > Thanks. I didn't make any special arrangements for FS or Skypiax to >> >>> > work on >> >>> > CentOS 5.3. I only enabled CentOS Plus yum repository and then >> >>> > install >> >>> > PAE >> >>> > kernel with following commands, >> >>> > >> >>> > root ~# yum update >> >>> > root ~# yum install kernel-PAE >> >>> > >> >>> > i installed PAE kernel just because i wanted to increase System RAM >> >>> > to >> >>> > 8GB >> >>> > before i deploy it for production use, so i can double or even >> >>> > triple >> >>> > Skypiax channels whenever i need so, without system or FS shutdown. >> >>> > >> >>> > As far as a skypiax configuration is concerned, i did modified >> >>> > mod_skypiax.c >> >>> > to add a couple of commands to dynamically add and remove Skypiax >> >>> > interfaces >> >>> > in a running FS process. However, this code does not replaces or >> >>> > changes any >> >>> > previous code. Other then that there is no significant change in >> >>> > configuration steps. Though i did use mod_xml_curl to dynamically >> >>> > update >> >>> > skypiax interface configuration in FS. >> >>> > >> >>> > >> >>> > Thank you. >> >>> > >> >>> > >> >>> > On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli >> >>> > >> >>> > wrote: >> >>> >> >> >>> >> Ciao Muhammad! >> >>> >> >> >>> >> What a good news! >> >>> >> >> >>> >> Centos is the most stable and performing platform for FS, so I >> >>> >> would >> >>> >> really love to test and document on the wiki how to have a stable >> >>> >> centos mod_skypiax installation. >> >>> >> >> >>> >> I'll find out your kernel ( Kernel >> >>> >> 2.6.18-92.1.22.el5.centos.plusPAE >> >>> >> ), and begin to test. In the mean time, do you have any hint, >> >>> >> special >> >>> >> procedure, etc you have done for having skypiax working well? >> >>> >> >> >>> >> Please, please, please let be in contact! :-) >> >>> >> >> >>> >> >> >>> >> Sincerely, >> >>> >> >> >>> >> Giovanni Maruzzelli >> >>> >> ========================================= >> >>> >> www.celliax.org >> >>> >> via Pierlombardo 9, 20135 Milano >> >>> >> Italy >> >>> >> gmaruzz at celliax dot org >> >>> >> Cell : +39-347-2665618 >> >>> >> Fax : +39-02-87390039 >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> On Wed, Jun 10, 2009 at 8:33 AM, Muhammad >> >>> >> Shahzad wrote: >> >>> >> > Sorry, i didn't visited the Jira link you mentioned. Now i know >> >>> >> > the >> >>> >> > issue >> >>> >> > and I have replied it there. >> >>> >> > >> >>> >> > Thank you. >> >>> >> > >> >>> >> > >> >>> >> > On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli >> >>> >> > >> >>> >> > wrote: >> >>> >> >> >> >>> >> >> Hi all, >> >>> >> >> >> >>> >> >> there are problems for mod_skypiax in recent centos, ?with more >> >>> >> >> than a >> >>> >> >> handful of concurrent Skype calls. >> >>> >> >> >> >>> >> >> Probably the problem is ALSA-related. >> >>> >> >> >> >>> >> >> Until it is solved, for production please use Ubuntu 8.04 (see >> >>> >> >> below), >> >>> >> >> some other Linux distro (and please write here your experience), >> >>> >> >> or >> >>> >> >> Windows. >> >>> >> >> >> >>> >> >> I modified the wiki page to reflect this ( >> >>> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk >> >>> >> >> ) >> >>> >> >> >> >>> >> >> If someone with CentOS knowledge can chime in I'll be grateful >> >>> >> >> :-). >> >>> >> >> >> >>> >> >> Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 >> >>> >> >> for >> >>> >> >> all infos, and feel free to contact me directly. >> >>> >> >> >> >>> >> >> -giovanni >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> Sincerely, >> >>> >> >> >> >>> >> >> Giovanni Maruzzelli >> >>> >> >> ========================================= >> >>> >> >> www.celliax.org >> >>> >> >> via Pierlombardo 9, 20135 Milano >> >>> >> >> Italy >> >>> >> >> gmaruzz at celliax dot org >> >>> >> >> Cell : +39-347-2665618 >> >>> >> >> Fax : +39-02-87390039 >> >>> >> >> >> >>> >> >> _______________________________________________ >> >>> >> >> Freeswitch-users mailing list >> >>> >> >> Freeswitch-users at lists.freeswitch.org >> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> >> http://www.freeswitch.org >> >>> >> > >> >>> >> > >> >>> >> > >> >>> >> > -- >> >>> >> > Muhammad Shahzad >> >>> >> > ----------------------------------- >> >>> >> > CISCO Rich Media Communication Specialist (CRMCS) >> >>> >> > CISCO Certified Network Associate (CCNA) >> >>> >> > Cell: +92 334 422 40 88 >> >>> >> > MSN: shari_786pk at hotmail.com >> >>> >> > Email: shaheryarkh at googlemail.com >> >>> >> > >> >>> >> > _______________________________________________ >> >>> >> > Freeswitch-users mailing list >> >>> >> > Freeswitch-users at lists.freeswitch.org >> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> > >> >>> >> > >> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> > http://www.freeswitch.org >> >>> >> > >> >>> >> > >> >>> >> >> >>> >> _______________________________________________ >> >>> >> Freeswitch-users mailing list >> >>> >> Freeswitch-users at lists.freeswitch.org >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >> >>> >> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> http://www.freeswitch.org >> >>> > >> >>> > >> >>> > >> >>> > -- >> >>> > Muhammad Shahzad >> >>> > ----------------------------------- >> >>> > CISCO Rich Media Communication Specialist (CRMCS) >> >>> > CISCO Certified Network Associate (CCNA) >> >>> > Cell: +92 334 422 40 88 >> >>> > MSN: shari_786pk at hotmail.com >> >>> > Email: shaheryarkh at googlemail.com >> >>> > >> >>> > _______________________________________________ >> >>> > Freeswitch-users mailing list >> >>> > Freeswitch-users at lists.freeswitch.org >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> > >> >>> > >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> > http://www.freeswitch.org >> >>> > >> >>> > >> >>> >> >>> _______________________________________________ >> >>> Freeswitch-dev mailing list >> >>> Freeswitch-dev at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> -- >> >> Muhammad Shahzad >> >> ----------------------------------- >> >> CISCO Rich Media Communication Specialist (CRMCS) >> >> CISCO Certified Network Associate (CCNA) >> >> Cell: +92 334 422 40 88 >> >> MSN: shari_786pk at hotmail.com >> >> Email: shaheryarkh at googlemail.com >> > >> > >> > >> > -- >> > Muhammad Shahzad >> > ----------------------------------- >> > CISCO Rich Media Communication Specialist (CRMCS) >> > CISCO Certified Network Associate (CCNA) >> > Cell: +92 334 422 40 88 >> > MSN: shari_786pk at hotmail.com >> > Email: shaheryarkh at googlemail.com >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jingwei.yang at gmail.com Wed Jun 10 00:33:36 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Wed, 10 Jun 2009 15:33:36 +0800 Subject: [Freeswitch-users] What's the right way to use skypiax with dialplan Message-ID: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> Hi All, I just finished installing freeSwitch and Skypiax. And I'm able to use skype api directly via the sk command like the following: freeswitch at localhost.localdomain>sk console skypiax1 freeswitch at localhost.localdomain>sk CALL userAAA It works like a charm and userAAA is able to receive the call and answer it. However, I'm stuck in figuring out the right way to use Skypiax with a dialplan. I've put a dialplan like below into /usr/local/freeswitch/conf/dialplan/default.xml On the freeswitch console, I'm not sure how to trigger this dialplan correctly. I've tried something like freeswitch at localhost.localdomain>originate sofia/external/root at 192.168.1.1002909 API CALL [originate(sofia/external/root at 192.168.1.100 2909)] output: -ERR MANDATORY_IE_MISSING freeswitch at localhost.localdomain>originate sofia/external/localdomain at localhost 2909 API CALL [originate(sofia/external/localdomain at localhost 2909)] output: -ERR NORMAL_TEMPORARY_FAILURE All failed with errors indicated above. Please let me know what's the right way to originate the call. Thanks! Regards, -Jingwei p.s. my os is CentOS 5.3. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/a183818c/attachment-0002.html From max.bridgewater at gmail.com Wed Jun 10 06:39:44 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 09:39:44 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls Message-ID: Hi, Getting to learn Freeswitch. So, please bear with me. Can somebody tells me how to do dial extension 1003 from the command line. I tried the following and the all failed originate 1003%192.168.10.103 originate 1003 at 192.168.10.103 originate 1003%192.168.10.103 & park() originate 1003 at 192.168.10.103 & park() Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/3171ea07/attachment-0002.html From dujinfang at gmail.com Wed Jun 10 07:15:58 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 22:15:58 +0800 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems In-Reply-To: <7b197bef0906100628y25948a97x59af0ebd02c47d71@mail.gmail.com> References: <7b197bef0906090945j6d79147bk55699082ed64cd5d@mail.gmail.com> <7b197bef0906100137o78dc6271lea7f86b24f64f3a1@mail.gmail.com> <7b197bef0906100247k26f082a9g2f7e26bf7f3499a3@mail.gmail.com> <7b197bef0906100628y25948a97x59af0ebd02c47d71@mail.gmail.com> Message-ID: <49EB1F70-BD3D-4CF7-837F-6DE8AC01D5D5@gmail.com> On Jun 10, 2009, at 9:28 PM, Giovanni Maruzzelli wrote: > On Wed, Jun 10, 2009 at 3:10 PM, dujinfang wrote: >> btw, I have changed the start scripts a bit to start X and skype >> separately, >> glad to share it if someone interested. > > I'm interested! :-) > ok in jira. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Wed Jun 10 07:21:03 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 22:21:03 +0800 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: Message-ID: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> originate sofia/default/1003 &echo() originate user/1003 &echo() originate user/1003 &park() On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote: > Hi, > > Getting to learn Freeswitch. So, please bear with me. Can somebody > tells me how to do dial extension 1003 from the command line. > I tried the following and the all failed > > originate 1003%192.168.10.103 > originate 1003 at 192.168.10.103 > originate 1003%192.168.10.103 & park() > originate 1003 at 192.168.10.103 & park() > > Thanks, > Max. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/831d2410/attachment-0002.html From dujinfang at gmail.com Wed Jun 10 07:25:43 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 10 Jun 2009 22:25:43 +0800 Subject: [Freeswitch-users] What's the right way to use skypiax with dialplan In-Reply-To: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> References: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> Message-ID: <8210C83F-5ADD-4291-B807-AE2877EE3F6C@gmail.com> On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote: > Hi All, > > I just finished installing freeSwitch and Skypiax. And I'm able to > use skype api directly via the sk command like the following: > > freeswitch at localhost.localdomain>sk console skypiax1 > freeswitch at localhost.localdomain>sk CALL userAAA > > It works like a charm and userAAA is able to receive the call and > answer it. However, I'm stuck in figuring out the right way to use > Skypiax with a dialplan. I've put a dialplan like below into /usr/ > local/freeswitch/conf/dialplan/default.xml > > > > > > > > On the freeswitch console, I'm not sure how to trigger this dialplan > correctly. I've tried something like > > freeswitch at localhost.localdomain>originate sofia/external/root at 192.168.1.100 > 2909 > API CALL [originate(sofia/external/root at 192.168.1.100 2909)] output: > -ERR MANDATORY_IE_MISSING the problem is the dial string not the dialplan I think, why not try originate skypiax/ANY/userBBB 2909 it should call userBBB and bridge to userAAA. > > > freeswitch at localhost.localdomain>originate sofia/external/ > localdomain at localhost 2909 > API CALL [originate(sofia/external/localdomain at localhost 2909)] > output: > -ERR NORMAL_TEMPORARY_FAILURE > > > All failed with errors indicated above. Please let me know what's > the right way to originate the call. Thanks! > > Regards, > -Jingwei > > p.s. my os is CentOS 5.3. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/e5368126/attachment-0002.html From larclap at yahoo.com Wed Jun 10 07:51:42 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 10 Jun 2009 07:51:42 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> Message-ID: <005301c9e9da$f8397ff0$e8ac7fd0$@com> There was no 0.0.0.0 anywhere. I used vi. I'll rotate the logs and restart FS without nc later today and report back. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 6:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/94abec9e/attachment-0002.html From max.bridgewater at gmail.com Wed Jun 10 07:54:41 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 10:54:41 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: Thanks, the first variant doesn't work for me. Any idea? I changed it to: originate sofia/internal/1003 at 192.168.10.103 & park() then i get the error message: "Cannot blind transfer 1 legged call". Also going through the wiki i discovered the concept of socket event and included the following extension under /usr/local/freeswitch/conf/dialplan/public/mysockt.xml: Can somebody gives me an example of dial string that would allow my call to be sent to this server socket? How would i place such a call using SJPhone? Best regards, Max. On Wed, Jun 10, 2009 at 10:21 AM, dujinfang wrote: > originate sofia/default/1003 &echo()originate user/1003 &echo() > originate user/1003 &park() > > > On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote: > > Hi, > > Getting to learn Freeswitch. So, please bear with me. Can somebody tells me > how to do dial extension 1003 from the command line. > I tried the following and the all failed > > originate 1003%192.168.10.103 > originate 1003 at 192.168.10.103 > originate 1003%192.168.10.103 & park() > originate 1003 at 192.168.10.103 & park() > > Thanks, > Max. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7c339943/attachment-0002.html From mike at jerris.com Wed Jun 10 08:07:48 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Jun 2009 11:07:48 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: <36AC81BC-AFE5-4D38-901B-C293535C52CC@jerris.com> the default alias was removed from the default configs last week, so new configs don't have this anymore. On Jun 10, 2009, at 10:54 AM, Max Bridgewater wrote: > Thanks, > > the first variant doesn't work for me. Any idea? > I changed it to: > > originate sofia/internal/1003 at 192.168.10.103 & park() > > then i get the error message: "Cannot blind transfer 1 legged call". > > > > Also going through the wiki i discovered the concept of socket event > and included the following extension under /usr/local/freeswitch/ > conf/dialplan/public/mysockt.xml: > > > > break="on-true"> > > > > > > Can somebody gives me an example of dial string that would allow my > call to be sent to this server socket? How would i place such a call > using SJPhone? > > Best regards, > Max. > > On Wed, Jun 10, 2009 at 10:21 AM, dujinfang > wrote: > originate sofia/default/1003 &echo() > originate user/1003 &echo() > originate user/1003 &park() > > > On Jun 10, 2009, at 9:39 PM, Max Bridgewater wrote: >> Hi, >> >> Getting to learn Freeswitch. So, please bear with me. Can somebody >> tells me how to do dial extension 1003 from the command line. >> I tried the following and the all failed >> >> originate 1003%192.168.10.103 >> originate 1003 at 192.168.10.103 >> originate 1003%192.168.10.103 & park() >> originate 1003 at 192.168.10.103 & park() >> >> Thanks, >> Max. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7d80b76b/attachment-0002.html From brian at freeswitch.org Wed Jun 10 08:11:25 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 10:11:25 -0500 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings If you're calling a locally registered user... you need to use user/ user at domain which uses dial-string from the params on the user or directory. Or sofia/profile/user%domain /b On Jun 10, 2009, at 9:54 AM, Max Bridgewater wrote: > Thanks, > > the first variant doesn't work for me. Any idea? > I changed it to: > > originate sofia/internal/1003 at 192.168.10.103 & park() > > then i get the error message: "Cannot blind transfer 1 legged call". > > > > Also going through the wiki i discovered the concept of socket event > and included the following extension under /usr/local/freeswitch/ > conf/dialplan/public/mysockt.xml: > > > > break="on-true"> > > > > > > Can somebody gives me an example of dial string that would allow my > call to be sent to this server socket? How would i place such a call > using SJPhone? > > Best regards, > Max. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/1b2a2f1a/attachment-0002.html From max.bridgewater at gmail.com Wed Jun 10 09:18:49 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 12:18:49 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: Thanks Folks; I'm making progress. The following origination string does make my non-registered SJPhone ring: {origination_caller_id_number=2000}sofia/external/some at 192.168.50.67 But why isn't it caught by the following extension? On Wed, Jun 10, 2009 at 11:11 AM, Brian West wrote: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > If you're calling a locally registered user... you need to use > user/user at domain which uses dial-string from the params on the user or > directory. Or sofia/profile/user%domain > > /b > > > On Jun 10, 2009, at 9:54 AM, Max Bridgewater wrote: > > Thanks, > > the first variant doesn't work for me. Any idea? > I changed it to: > > originate sofia/internal/1003 at 192.168.10.103 & park() > > then i get the error message: "Cannot blind transfer 1 legged call". > > > > Also going through the wiki i discovered the concept of socket event and > included the following extension under > /usr/local/freeswitch/conf/dialplan/public/mysockt.xml: > > > > break="on-true"> > /> > > > > > Can somebody gives me an example of dial string that would allow my call to > be sent to this server socket? How would i place such a call using SJPhone? > > Best regards, > Max. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/7d58a043/attachment-0002.html From mike at jerris.com Wed Jun 10 09:52:51 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Jun 2009 12:52:51 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Dialplan_XML break="on-true" ? On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote: > Thanks Folks; I'm making progress. The following origination string > does make my non-registered SJPhone ring: > > {origination_caller_id_number=2000}sofia/external/some at 192.168.50.67 > > > But why isn't it caught by the following extension? > > > > > break="on-true"> > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/69f988cc/attachment-0002.html From paul at ringcarrier.com Wed Jun 10 10:02:16 2009 From: paul at ringcarrier.com (Paul Mahler) Date: Wed, 10 Jun 2009 10:02:16 -0700 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? Message-ID: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Hello Everybody, I have a large project coming up. I'm interested in using Freeswitch instead of SER and Asterisk. What is the current status of Freeswitch? Can I safely use it in a large scale commercial environment? How active is the Freeswitch developer community? I am concerned that Freeswitch doesn't seem to have gained much traction compared to Asterisk. How viable is the Freeswitch project? Please help me understand why I can safely shit can Asterisk and move to Freeswitch. Thank You, Paul _________________ Paul Mahler paul at ringcarrier.com From max.bridgewater at gmail.com Wed Jun 10 10:20:01 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Wed, 10 Jun 2009 13:20:01 -0400 Subject: [Freeswitch-users] Newbie Question wrt Originating calls In-Reply-To: References: <6CDC3B91-FE1D-467E-B5AD-76D938CB53F4@gmail.com> Message-ID: Well, i assume break="on-true" means, that if this extension is matched then execute its actions and stop there. That would correspond to what i'm trying to do. Anyway i removed this attribute and still nothing is being sent to the socket. Let me give you more context. This extension is put in a separate file AS IS under /usr/local/freeswitch/conf/dialplan/public/myextension.xml. Is this the right place? I have the impression that it is not seen or processed by Freeswitch. The latest thing i tried is to wrap the extension in a context element with name attribute "public". It did'nt help. Any clue? On Wed, Jun 10, 2009 at 12:52 PM, Michael Jerris wrote: > http://wiki.freeswitch.org/wiki/Dialplan_XML > break="on-true" ? > > On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote: > > Thanks Folks; I'm making progress. The following origination string does > make my non-registered SJPhone ring: > > {origination_caller_id_number=2000}sofia/external/some at 192.168.50.67 > > > But why isn't it caught by the following extension? > > > > > break="on-true"> > /> > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/b81b14cb/attachment-0002.html From benj at teliax.com Wed Jun 10 10:18:43 2009 From: benj at teliax.com (Ben Jones) Date: Wed, 10 Jun 2009 11:18:43 -0600 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Message-ID: <4A2FEAF3.1080807@teliax.com> Hi Paul, I'll tell you this. We (Teliax) made the decision to use FreeSWITCH instead of Asterisk in late 2008 and we haven't looked back since. We're a fairly large SIP/IAX provider with four POPs located throughout the US, each one running FS. The dev community surrounding FS is excellent, they're always helpful and if you check the changelogs (http://fisheye.freeswitch.org/changelog/FreeSWITCH/) you can clearly see there's all sorts of work going on daily. As a company with 5000+ customers with an emphasis on call completion and high quality, we're happy to say that we can rely on FreeSWITCH every single day to provide us with what Asterisk couldn't. Don't get me wrong, Asterisk is a good PBX (it's quick to deploy, relatively easy, and well supported) but try pushing thousands of calls through it at once without it choking. FreeSWITCH handles it with 94% processor idle. I realize I might not answer all your questions, but I just wanted to throw in our vote for you to replace Asterisk with FreeSWITCH. Maybe seeing a real-world example such as ours will help your decision to make the switch. (pun fully intended) Hope this helps, Ben J Paul Mahler wrote: > Hello Everybody, > > I have a large project coming up. I'm interested in using Freeswitch > instead of SER and Asterisk. > > What is the current status of Freeswitch? Can I safely use it in a > large scale commercial environment? How active is the Freeswitch > developer community? > > I am concerned that Freeswitch doesn't seem to have gained much > traction compared to Asterisk. How viable is the Freeswitch project? > > Please help me understand why I can safely shit can Asterisk and move > to Freeswitch. > > Thank You, > > Paul > _________________ > Paul Mahler > paul at ringcarrier.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From helmut.kuper at ewetel.de Wed Jun 10 10:32:41 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 10 Jun 2009 19:32:41 +0200 Subject: [Freeswitch-users] Database and "Too many open files" Problem Message-ID: <4A2FEE39.1030709@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I updated to latest trunk. Everything compiled well. FS starts up without errors. Calls are executed successfully. But after a few minutes with very few call activity I get this lines on console: 2009-06-10 18:21:16.52527 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) 2009-06-10 18:21:16.152501 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) 2009-06-10 18:21:16.252481 [ERR] switch_core_sqldb.c:95 SQL ERR [unable to open database file] (update tasks set task_sql_manager='' where task_id=1; ) ... 2009-06-10 18:22:37.121405 [CRIT] switch_core_sqldb.c:209 SQL thread unable to commit transaction, records lost! I modified switch_core_sqldb.c to output the actual sql statement as well to get an idea which database is affected. It's the core-db. Mixed with that lines above I got lines like this: 2009-06-10 18:21:32.656605 [ERR] mod_sndfile.c:194 Error Opening File [/opt/app/voip/ippbx/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav] [System error : Too many open files in system.] This may points to the ulimit thing, cause I run FS as non root (on centos 5.3). ulimit -a shows: [ippbx at ippbx-test-node0 ~]$ ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited /etc/security/limits.conf contains: ippbx - nofile 999999 ippbx - core unlimited ippbx - data unlimited ippbx - fsize unlimited ippbx - sigpending unlimited ippbx - msgqueue unlimited ippbx - nproc unlimited ippbx - locks unlimited So has anyone an idea what's wrong? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKL+454tZeNddg3dwRAqcCAJsFSEWuI2X1fvOF2rmIFoSmRdBZzQCggD8B MQiS7RZkOoL9lhV8LV+pL7s= =v7QY -----END PGP SIGNATURE----- From benj at teliax.com Wed Jun 10 10:34:04 2009 From: benj at teliax.com (Ben Jones) Date: Wed, 10 Jun 2009 11:34:04 -0600 Subject: [Freeswitch-users] Clarification on start_dtmf_generate? Message-ID: <4A2FEE8C.8090906@teliax.com> Hello all, The documentation has this to say on start_dtmf_generate: As an example, Placing this in the Dialplan prior to bridging a call, will allow a phone set to rfc2833 ( info ) to send DTMF tones (in-band ) out to the recipient (IVR) or Auto Attendants. Thus changing the outgoing routing from (info) to (in-band). From what I understand rfc2833 != info. Does this mean if a user is set for rfc2833 OR info that FS will generate inband tones to send out? Does it only generate tones for info? If I can get any clarification on this I'll be more than happy to update the documentation. Thanks, Ben J -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From brian at freeswitch.org Wed Jun 10 10:40:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 12:40:58 -0500 Subject: [Freeswitch-users] Clarification on start_dtmf_generate? In-Reply-To: <4A2FEE8C.8090906@teliax.com> References: <4A2FEE8C.8090906@teliax.com> Message-ID: <72CBB468-4382-4F09-B1F7-9CC678F61243@freeswitch.org> Yes. /b On Jun 10, 2009, at 12:34 PM, Ben Jones wrote: > Does this mean if a user is set for rfc2833 OR info that FS will > generate inband tones to send out? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/5e5c8d3b/attachment-0002.html From freeswitch-users at digitaldan.com Wed Jun 10 10:57:16 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 10 Jun 2009 11:57:16 -0600 (MDT) Subject: [Freeswitch-users] Recording time limit? In-Reply-To: <87f2f3b90906091506s5da8b925u256eaeaaf8c0fbb3@mail.gmail.com> Message-ID: <9943583.12931244656630422.JavaMail.daniel@radio> Yep, we tracked it down to the originating pbx ( a cisco call manager) which had a 12 hour limit on outbound calls, thanks for your help. D- ----- Original Message ----- From: "Michael Collins" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 9, 2009 4:06:01 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording time limit? On Tue, Jun 9, 2009 at 2:55 PM, Brian West < brian at freeswitch.org > wrote: Someone hung the call up thats all I can imagine. Its not going to be us but maybe the far side you're talking to. Look at the sip trace bet the far end is sending you a BYE This would make sense. Some systems will have a hard limit on the length of a call and will disconnect automatically because they interpret a 12 hour call as a "problem" and the "solution" is to hang up. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/df7f1bd8/attachment-0002.html From msc at freeswitch.org Wed Jun 10 11:04:37 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 10 Jun 2009 11:04:37 -0700 Subject: [Freeswitch-users] Clarification on start_dtmf_generate? In-Reply-To: <72CBB468-4382-4F09-B1F7-9CC678F61243@freeswitch.org> References: <4A2FEE8C.8090906@teliax.com> <72CBB468-4382-4F09-B1F7-9CC678F61243@freeswitch.org> Message-ID: I will clarify the point on the wiki since it is a bit inaccurate -MC Sent from my iPhone On Jun 10, 2009, at 10:40 AM, Brian West wrote: > Yes. > > /b > > On Jun 10, 2009, at 12:34 PM, Ben Jones wrote: > >> Does this mean if a user is set for rfc2833 OR info that FS will >> generate inband tones to send out? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/bf357658/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Jun 10 11:30:55 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 10 Jun 2009 19:30:55 +0100 Subject: [Freeswitch-users] Problems with make current Message-ID: Hi Guys, Ran make current today, and am getting the following errors. I ran bootstrap and configure, but still get these messages. Any ideas ? Looks like I'm now missing some libraries Regards, configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/f9ec3533/attachment-0002.html From brian at freeswitch.org Wed Jun 10 11:36:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 13:36:01 -0500 Subject: [Freeswitch-users] Problems with make current In-Reply-To: References: Message-ID: <3C17ED7A-A837-45D3-8469-3CDDABF9EB83@freeswitch.org> need to rebootstrap. /b On Jun 10, 2009, at 1:30 PM, Nik Middleton wrote: > Hi Guys, > > Ran make current today, and am getting the following errors. I ran > bootstrap and configure, but still get these messages. > > Any ideas ? Looks like I?m now missing some libraries > > Regards, > > configure: configuring in libs/pcre > configure: running /bin/sh './configure.gnu' --prefix=/usr/local/ > freeswitch --cache-file=/dev/null --srcdir=. > configure: error: cannot find sources (pcre.h.in) in . > configure: error: /bin/sh './configure.gnu' failed for libs/pcre Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/020f5c63/attachment-0002.html From mike at jerris.com Wed Jun 10 11:50:07 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 10 Jun 2009 14:50:07 -0400 Subject: [Freeswitch-users] Problems with make current In-Reply-To: References: Message-ID: <9B3365B3-9C75-412D-B3CF-623F32F26B8C@jerris.com> your svn update failed, rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make current On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote: > Hi Guys, > > Ran make current today, and am getting the following errors. I ran > bootstrap and configure, but still get these messages. > > Any ideas ? Looks like I?m now missing some libraries > > Regards, > > configure: configuring in libs/pcre > configure: running /bin/sh './configure.gnu' --prefix=/usr/local/ > freeswitch --cache-file=/dev/null --srcdir=. > configure: error: cannot find sources (pcre.h.in) in . > configure: error: /bin/sh './configure.gnu' failed for libs/pcre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/5fcb649c/attachment-0002.html From marketing at cluecon.com Wed Jun 10 13:11:53 2009 From: marketing at cluecon.com (Michael Collins) Date: Wed, 10 Jun 2009 13:11:53 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Important Reminders Message-ID: <87f2f3b90906101311g4e41c324p279918f855703e6b@mail.gmail.com> Friends, ClueCon 2009 is fast approaching! We are definitely looking forward to seeing everyone in Chicago this August. If you haven't finalized your plans to attend, please do so right away. Time is running out! The early-bird registration special of $499 per person will expire at the end of June - only 20 days away. Also, registration as a whole ends on July 21st, so don't delay. Lastly, be sure to book your room at the Chicago Wyndham. Try expedia.com to see what kind of deals are still available. Please call 877.742.CLUE to get registered today! -The ClueCon Team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/33cd63ad/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Jun 10 13:53:31 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 10 Jun 2009 21:53:31 +0100 Subject: [Freeswitch-users] Problems with make current In-Reply-To: <9B3365B3-9C75-412D-B3CF-623F32F26B8C@jerris.com> References: <9B3365B3-9C75-412D-B3CF-623F32F26B8C@jerris.com> Message-ID: Thanks, that did the trick Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 10 June 2009 19:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with make current your svn update failed, rm -rf libs/pcre && svn update && ./bootstrap.sh && ./configure && make current On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote: Hi Guys, Ran make current today, and am getting the following errors. I ran bootstrap and configure, but still get these messages. Any ideas ? Looks like I'm now missing some libraries Regards, configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/d1c14146/attachment-0002.html From larclap at yahoo.com Wed Jun 10 14:04:08 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 10 Jun 2009 14:04:08 -0700 Subject: [Freeswitch-users] Remove example.com gateway? Message-ID: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> Is it OK to remove the example.com gateway? I removed the example.xml files in sip_profiles/external and sip_profiles/internal and changed the default_provider from example.com to myprovider.com. But I still see myprovider.com as a gateway in sofia status. How do I get rid of this, of course, if it's OK. I want to simplify the configuration. I dialed 18885551212 and it went into [local.example.com]; and 8885551212 it went into [eavesdrop]. I think I'm talking more than just the gateway, but I do want to make the dialplan work in the way I expect. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/f2cef6bd/attachment-0002.html From brian at freeswitch.org Wed Jun 10 14:10:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 16:10:01 -0500 Subject: [Freeswitch-users] Remove example.com gateway? In-Reply-To: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> References: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> Message-ID: You can remove it all you want but those settings in vars.xml won't do anything because its expanded in conf/directory/default/example.com.xml /b On Jun 10, 2009, at 4:04 PM, Lars Zeb wrote: > Is it OK to remove the example.com gateway? I removed the > example.xml files in sip_profiles/external and sip_profiles/internal > and changed the default_provider from example.com to myprovider.com. > > But I still see myprovider.com as a gateway in sofia status. How do > I get rid of this, of course, if it?s OK. > > I want to simplify the configuration. I dialed 18885551212 and it > went into [local.example.com]; and 8885551212 it went into > [eavesdrop]. I think I?m talking more than just the gateway, but I > do want to make the dialplan work in the way I expect. > > Thanks, Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/96a513d8/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 10 14:14:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 10 Jun 2009 16:14:26 -0500 Subject: [Freeswitch-users] Database and "Too many open files" Problem In-Reply-To: <4A2FEE39.1030709@ewetel.de> References: <4A2FEE39.1030709@ewetel.de> Message-ID: <191c3a030906101414j1107e49bk3aa8fb778454451c@mail.gmail.com> This is a bug in the latest wanpipe. you can wait for the next release of wanpipe/openzap or you can downgrade to wanpipe 4.1 and rebuild FS On Wed, Jun 10, 2009 at 12:32 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > today I updated to latest trunk. Everything compiled well. FS starts up > without errors. Calls are executed successfully. > > But after a few minutes with very few call activity I get this lines on > console: > > > 2009-06-10 18:21:16.52527 [ERR] switch_core_sqldb.c:95 SQL ERR [unable > to open database file] (update tasks set task_sql_manager='' where > task_id=1; > ) > 2009-06-10 18:21:16.152501 [ERR] switch_core_sqldb.c:95 SQL ERR [unable > to open database file] (update tasks set task_sql_manager='' where > task_id=1; > ) > 2009-06-10 18:21:16.252481 [ERR] switch_core_sqldb.c:95 SQL ERR [unable > to open database file] (update tasks set task_sql_manager='' where > task_id=1; > ) > > ... > > 2009-06-10 18:22:37.121405 [CRIT] switch_core_sqldb.c:209 SQL thread > unable to commit transaction, records lost! > > I modified switch_core_sqldb.c to output the actual sql statement as > well to get an idea which database is affected. It's the core-db. > > > > Mixed with that lines above I got lines like this: > > 2009-06-10 18:21:32.656605 [ERR] mod_sndfile.c:194 Error Opening File > > [/opt/app/voip/ippbx/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav] > [System error : Too many open files in system.] > > > This may points to the ulimit thing, cause I run FS as non root (on > centos 5.3). > > ulimit -a shows: > [ippbx at ippbx-test-node0 ~]$ ulimit -a > core file size (blocks, -c) 0 > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) 32 > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 10240 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > /etc/security/limits.conf contains: > ippbx - nofile 999999 > ippbx - core unlimited > ippbx - data unlimited > ippbx - fsize unlimited > ippbx - sigpending unlimited > ippbx - msgqueue unlimited > ippbx - nproc unlimited > ippbx - locks unlimited > > > So has anyone an idea what's wrong? > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKL+454tZeNddg3dwRAqcCAJsFSEWuI2X1fvOF2rmIFoSmRdBZzQCggD8B > MQiS7RZkOoL9lhV8LV+pL7s= > =v7QY > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/a8983233/attachment-0002.html From msc at freeswitch.org Wed Jun 10 14:51:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Jun 2009 14:51:00 -0700 Subject: [Freeswitch-users] Remove example.com gateway? In-Reply-To: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> References: <010701c9ea0e$ff24bfd0$fd6e3f70$@com> Message-ID: <87f2f3b90906101451p3af74a65ta52da2646954500a@mail.gmail.com> On Wed, Jun 10, 2009 at 2:04 PM, Lars Zeb wrote: > Is it OK to remove the example.com gateway? I removed the example.xml > files in sip_profiles/external and sip_profiles/internal and changed the > default_provider from example.com to myprovider.com. > > If you just want it to "go away" then you can always rename the file to something like 'example.com.noload' -MC > > > But I still see myprovider.com as a gateway in sofia status. How do I get > rid of this, of course, if it?s OK. > > > > I want to simplify the configuration. I dialed 18885551212 and it went into > [local.example.com]; and 8885551212 it went into [eavesdrop]. I think I?m > talking more than just the gateway, but I do want to make the dialplan work > in the way I expect. > > > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/2bc635a4/attachment-0002.html From msc at freeswitch.org Wed Jun 10 15:34:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Jun 2009 15:34:52 -0700 Subject: [Freeswitch-users] A Few New Blog Posts Message-ID: <87f2f3b90906101534q52d4ad06l9524fc57c10a9407@mail.gmail.com> FYI, I've added a few new posts on the main FreeSWITCH page: http://www.freeswitch.org/node/190 - OpenSimulator on EC2 http://www.freeswitch.org/node/191 - Rob Smart FS as-home-PBX (U.K.) how-to Enjoy! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/798e0975/attachment-0002.html From palani.sivagurunathan at gmail.com Wed Jun 10 15:45:08 2009 From: palani.sivagurunathan at gmail.com (palani vel) Date: Wed, 10 Jun 2009 18:45:08 -0400 Subject: [Freeswitch-users] palani vel sent you a Friend Request on Yaari Message-ID: palani vel wants you to join Yaari! Is palani your friend? Yes, palani is my friend! No, palani isn't my friend. Please respond or palani may think you said no :( Thanks, The Yaari Team ----------------------------------------------------------- Yaari Inc., 358 Angier Ave NE Atlanta, GA 30312 Privacy Policy | Unsubscribe | Terms of Service YaariNYX927AYT966XXO323IQT265 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/0f0eec8f/attachment-0002.html From larclap at yahoo.com Wed Jun 10 16:54:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 10 Jun 2009 16:54:38 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: References: <005001c9e8a9$76c5c780$64515680$@com> <012501c9e91b$cf654880$6e2fd980$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> Message-ID: <015b01c9ea26$d0abb2e0$720318a0$@com> Rupa, I think the console log has information in it that log/freeswitch.log does not. Console: [root at fs bin]# ./freeswitch 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (Invalid or incomplete multibyte or wide character) 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! log/freeswitch.log: 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock interface 'console' to wait for existing references. (from previous Freeswitch invocation) 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding Dialplan 'enum' 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding Application 'enum' 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum' 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum_auto' 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default template. 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template example. 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template linksys. 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template asterisk. I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console log. However, the disk log begins at 16:12:58, whereas the console log starts at 16:12:50. The console log finishes its NAT and UPnP reporting before the disk log begins, so I wouldn't see any 0.0.0.0 if it were present. The [ERR] was due to me removing example.xml from sip_profiles/internal. I put it back after this. I don't understand the following command in conf/sofia.conf.xml. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 6:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/5d6b3763/attachment-0002.html From msc at freeswitch.org Wed Jun 10 17:18:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Jun 2009 17:18:10 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015b01c9ea26$d0abb2e0$720318a0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> <015b01c9ea26$d0abb2e0$720318a0$@com> Message-ID: <87f2f3b90906101718g72911b8fv326ee9c470fdf097@mail.gmail.com> On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb wrote: > Rupa, > > > > I think the console log has information in it that log/freeswitch.log does > not. > > > > Console: > > [root at fs bin]# ../freeswitch > > 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing > Engine. > > 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch > thread 0 > > 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (No such file or directory) > > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (Invalid or incomplete multibyte or wide character) > > 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT > > 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 > > 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 > > 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 > > 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 > > 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 > > 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP > > 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT > detected! > > > > log/freeswitch.log: > > 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock > interface 'console' to wait for existing references. (from previous > Freeswitch invocation) > > 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding > Dialplan 'enum' > > 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding > Application 'enum' > > 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum' > > 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum_auto' > > 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default > template. > > 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. > > 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. > > 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template > example. > > 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. > > 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template > linksys. > > 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template > asterisk. > > > > I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console > log. However, the disk log begins at 16:12:58, whereas the console log > starts at 16:12:50. The console log finishes its NAT and UPnP reporting > before the disk log begins, so I wouldn?t see any 0.0.0.0 if it were > present. > > > > The [ERR] was due to me removing example.xml from sip_profiles/internal. I > put it back after this. I don?t understand the following command in > conf/sofia.conf.xml. > > > > > > I think this is just a cosmetic error. You could probably put an empty xml file in sip_profiles/internal and be done with it. Or possibly have just an empty include node, like "" Try it out and report back - we're dying to know what happens! ;) -MC > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 6:05 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > if you haven't changed your logging, then it is probably ok. The 0.0.0.0 > thing is logged at error level, so will show up in the logs. How did you > search? Grep? > > grep '0\.0\.0\.0' freeswitch.log > > On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: > > Rupa, > > > > What options do I have for setting up logging? I?m sorry, but I don?t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 5:19 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > > > On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > > You might want to review how you have your logging setup then. The example > I gave you was copied/pasted out of my freeswitch.log file while testing > this fix. > > > -- > -Rupa > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/245ee933/attachment-0002.html From john at feith.com Wed Jun 10 17:31:53 2009 From: john at feith.com (John Wehle) Date: Wed, 10 Jun 2009 20:31:53 -0400 (EDT) Subject: [Freeswitch-users] Finding all active calls belonging to the same phone Message-ID: <200906110031.n5B0Vr7S006625@jwlab.FEITH.COM> To duplicate our old PBX park functionality I need for a user who's on a call to be able to pick up a second line and dial a number to park the other call which is on his phone. I have something working, however am curious if there's a better way to accomplish this. Specifically I'm curious if there's a recommended way to find all the calls to / from the same phone / channel. I ended up configuring *5 to call a javascript program which: a) Gets the uuid from the session and uses it to search "show channels" to find the channel name. b) Normalizes the channel name and uses it to search "show channels" to find an uuid associated with the channel which is different from the one that invoked *5. c) Uses the uuid from "b" to search "show calls" to find the peer uuid. d) Uses uuid_setvar to set hangup_after_bridge=false and uuid_transfer to transfer the peer uuid to the proper fifo. One of the problems I ran into is the channel name has slightly different formats depending on whether it is an inbound or outbound channel. E.g.: sofia/internal/1003 at XXX.XXX.XXX.XXX sofia/internal/1003 at XXX.XXX.XXX.XXX:5060 sofia/internal/sip:1003 at YYY.YYY.YYY.YYY:5060;transport=udp;... where XXX is the freeswitch box and YYY is the phone. I created the following function to normalize the channel name for comparison: function normalize_channel_name (name, direction, ip_addr) { var re = /^sofia\//g; var length = name.search (re); var new_name = name; if (length == -1) return new_name; if (direction == "inbound") { re = /@.*$/g; new_name = name.replace (re, "@" + ip_addr); } else if (direction == "outbound") { re = /\/sip:(.*@[^:]*):.*$/g; new_name = name.replace (re, "/$1"); } return new_name; } Suggestions for a better approach? Keep in mind that my existing user population expects (for better or worse) to use *5 to park the call on their phone so I'm somewhat limited in what I can do. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From jcromes at gmail.com Wed Jun 10 17:49:01 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Wed, 10 Jun 2009 19:49:01 -0500 Subject: [Freeswitch-users] Web page thoughts Message-ID: <4A30547D.4090101@gmail.com> My company is currently investigating a couple of projects that may take me in the direction of FreeSwitch... In general, our management does not often consider open source software for projects such as this, but I've been successful in proving to them recently that open source can deliver. FreeSwitch is a *very* professional and polished product, I can tell - from the code and from the community. Unfortunately, I've been hesitant to send people to your webpage lately because it went downhill a few weeks ago. Whenever I think about one of our executives going to your webpage (after my recommendation) and seeing a picture of people clanking beer glasses, or some idiot tied up in phone cables, I cringe. I know you're advertising for ClueCon, but honestly, some of those huge images on your front page really knock your product down a peg in professionalism. Anyway, I'm pretty new to the community and I don't claim to be a web designer. You have an excellent piece of software, but if I didn't already know that about FreeSwitch, your webpage would not make a good first impression. Please take that for what it's worth... I wanted to voice my opinion because if I'm thinking it, others may be as well. Thoughts anyone? J (Have I mentioned how awesome your source browser is though??!!) From rupa at rupa.com Wed Jun 10 17:58:30 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 10 Jun 2009 19:58:30 -0500 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <015b01c9ea26$d0abb2e0$720318a0$@com> References: <005001c9e8a9$76c5c780$64515680$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> <015b01c9ea26$d0abb2e0$720318a0$@com> Message-ID: Oh, it is probable that logging is initialized after nat.... so the first pass won't show up in the filesystem logs. It would should up on subsequent nat initialization (what I was testing with some new code). On Wed, Jun 10, 2009 at 6:54 PM, Lars Zeb wrote: > Rupa, > > > > I think the console log has information in it that log/freeswitch.log does > not. > > > > Console: > > [root at fs bin]# ../freeswitch > > 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing > Engine. > > 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch > thread 0 > > 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (No such file or directory) > > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml > (Invalid or incomplete multibyte or wide character) > > 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT > > 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 > > 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 > > 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 > > 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 > > 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 > > 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP > > 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT > detected! > > > > log/freeswitch.log: > > 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock > interface 'console' to wait for existing references. (from previous > Freeswitch invocation) > > 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding > Dialplan 'enum' > > 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding > Application 'enum' > > 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum' > > 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'enum_auto' > > 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default > template. > > 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. > > 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. > > 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template > example. > > 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. > > 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template > linksys. > > 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template > asterisk. > > > > I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console > log. However, the disk log begins at 16:12:58, whereas the console log > starts at 16:12:50. The console log finishes its NAT and UPnP reporting > before the disk log begins, so I wouldn?t see any 0.0.0.0 if it were > present. > > > > The [ERR] was due to me removing example.xml from sip_profiles/internal. I > put it back after this. I don?t understand the following command in > conf/sofia.conf.xml. > > > > > > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 6:05 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > if you haven't changed your logging, then it is probably ok. The 0.0.0.0 > thing is logged at error level, so will show up in the logs. How did you > search? Grep? > > grep '0\.0\.0\.0' freeswitch.log > > On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: > > Rupa, > > > > What options do I have for setting up logging? I?m sorry, but I don?t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 09, 2009 5:19 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Can't hear outbound calls > > > > > > On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > > You might want to review how you have your logging setup then. The example > I gave you was copied/pasted out of my freeswitch.log file while testing > this fix. > > > -- > -Rupa > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/88df1ec5/attachment-0002.html From jingwei.yang at gmail.com Wed Jun 10 18:25:21 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 11 Jun 2009 09:25:21 +0800 Subject: [Freeswitch-users] What's the right way to use skypiax with dialplan In-Reply-To: <8210C83F-5ADD-4291-B807-AE2877EE3F6C@gmail.com> References: <13529f9d0906100033r52918f0ew94bc52cf8890beb2@mail.gmail.com> <8210C83F-5ADD-4291-B807-AE2877EE3F6C@gmail.com> Message-ID: <13529f9d0906101825y76be738bhf957e8b30d829b94@mail.gmail.com> Yes, it's the right way to go! Thanks, man! On Wed, Jun 10, 2009 at 10:25 PM, dujinfang wrote: > > On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote: > > Hi All, > > I just finished installing freeSwitch and Skypiax. And I'm able to use > skype api directly via the sk command like the following: > > freeswitch at localhost.localdomain>sk console skypiax1 > freeswitch at localhost.localdomain>sk CALL userAAA > > It works like a charm and userAAA is able to receive the call and answer > it. However, I'm stuck in figuring out the right way to use Skypiax with a > dialplan. I've put a dialplan like below into > /usr/local/freeswitch/conf/dialplan/default.xml > > > > > > > > On the freeswitch console, I'm not sure how to trigger this dialplan > correctly. I've tried something like > > freeswitch at localhost.localdomain>originate sofia/external/ > root at 192.168.1.100 2909 > API CALL [originate(sofia/external/root at 192.168.1.100 2909)] output: > -ERR MANDATORY_IE_MISSING > > > the problem is the dial string not the dialplan I think, why not try > > originate skypiax/ANY/userBBB 2909 > > it should call userBBB and bridge to userAAA. > > > > freeswitch at localhost.localdomain>originate > sofia/external/localdomain at localhost 2909 > API CALL [originate(sofia/external/localdomain at localhost 2909)] output: > -ERR NORMAL_TEMPORARY_FAILURE > > > All failed with errors indicated above. Please let me know what's the right > way to originate the call. Thanks! > > Regards, > -Jingwei > > p.s. my os is CentOS 5.3. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/f1c89438/attachment-0002.html From diego.viola at gmail.com Wed Jun 10 18:39:24 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 10 Jun 2009 21:39:24 -0400 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <20090609033051.GA28848@jdc.jasonjgw.net> References: <005001c9e8a9$76c5c780$64515680$@com> <20090609033051.GA28848@jdc.jasonjgw.net> Message-ID: <86a32abc0906101839n67119d0fs935401a4a45e5f32@mail.gmail.com> Another vote for Git here :). On Mon, Jun 8, 2009 at 11:30 PM, Jason White wrote: > Lars Zeb wrote: > > I had a working FS installation which I messed up by doing a fresh > install. > > I tried to integrate all my custom changes, but I'm sure I screwed > something > > up. > > Git is an excellent tool for keeping track of FreeSWITCH configuration > changes. The history of my configuration is maintained in a git repository > under /opt/freeswitch/conf - git simply creates a .git subdirectory to > store > all of the revisions as they are committed. > > Git revert and git stash have been very useful at times, not to mention git > reset --hard. > > Since Git is used for Linux kernel development, it should be available from > most recent Linux distributions, and it can probably be compiled for other > Unix-like environments as well. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/1819885a/attachment-0002.html From brian at freeswitch.org Wed Jun 10 19:29:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 10 Jun 2009 21:29:49 -0500 Subject: [Freeswitch-users] Finding all active calls belonging to the same phone In-Reply-To: <200906110031.n5B0Vr7S006625@jwlab.FEITH.COM> References: <200906110031.n5B0Vr7S006625@jwlab.FEITH.COM> Message-ID: Why not use the presence events to keep that state? /b On Jun 10, 2009, at 7:31 PM, John Wehle wrote: > To duplicate our old PBX park functionality I need for a user who's > on a call to be able to pick up a second line and dial a number to > park the other call which is on his phone. I have something working, > however am curious if there's a better way to accomplish this. > > Specifically I'm curious if there's a recommended way to find all the > calls to / from the same phone / channel. > > I ended up configuring *5 to call a javascript program which: > > a) Gets the uuid from the session and uses it to search "show > channels" > to find the channel name. > > b) Normalizes the channel name and uses it to search "show channels" > to find an uuid associated with the channel which is different > from > the one that invoked *5. > > c) Uses the uuid from "b" to search "show calls" to find the peer > uuid. > > d) Uses uuid_setvar to set hangup_after_bridge=false and > uuid_transfer > to transfer the peer uuid to the proper fifo. > > One of the problems I ran into is the channel name has slightly > different > formats depending on whether it is an inbound or outbound channel. > E.g.: > > sofia/internal/1003 at XXX.XXX.XXX.XXX > sofia/internal/1003 at XXX.XXX.XXX.XXX:5060 > sofia/internal/sip:1003 at YYY.YYY.YYY.YYY:5060;transport=udp;... > > where XXX is the freeswitch box and YYY is the phone. I created the > following function to normalize the channel name for comparison: > > function normalize_channel_name (name, direction, ip_addr) > { > var re = /^sofia\//g; > var length = name.search (re); > var new_name = name; > > if (length == -1) > return new_name; > > if (direction == "inbound") { > re = /@.*$/g; > > new_name = name.replace (re, "@" + ip_addr); > } > else if (direction == "outbound") { > re = /\/sip:(.*@[^:]*):.*$/g; > > new_name = name.replace (re, "/$1"); > } > > return new_name; > } > > Suggestions for a better approach? Keep in mind that my existing user > population expects (for better or worse) to use *5 to park the call on > their phone so I'm somewhat limited in what I can do. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: > john at feith.com | > | John Wehle | Fax: 1-215-540-5495 > | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From johnd at defyne.org Wed Jun 10 20:04:22 2009 From: johnd at defyne.org (John Dalgliesh) Date: Thu, 11 Jun 2009 13:04:22 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques Message-ID: Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John From jason at voicesession.com Wed Jun 10 20:09:48 2009 From: jason at voicesession.com (lee jason) Date: Thu, 11 Jun 2009 11:09:48 +0800 Subject: [Freeswitch-users] SIP Registration Address Message-ID: <2cbf225c0906102009vf36fe52gf6ebd8d0c704d44b@mail.gmail.com> Dear All, I just have a question, How can I use Freeswitch to blind two IP address for SIP registration at same port(5060 UDP)? Thanks a lot. Jason Lee > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/2c3ede67/attachment-0002.html From mrene_lists at avgs.ca Wed Jun 10 20:12:19 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 10 Jun 2009 23:12:19 -0400 Subject: [Freeswitch-users] SIP Registration Address In-Reply-To: <2cbf225c0906102009vf36fe52gf6ebd8d0c704d44b@mail.gmail.com> References: <2cbf225c0906102009vf36fe52gf6ebd8d0c704d44b@mail.gmail.com> Message-ID: Hi, Create two sip profiles, one per IP. Math On 10-Jun-09, at 11:09 PM, lee jason wrote: > Dear All, > > I just have a question, How can I use Freeswitch to blind two > IP address for SIP registration at same port(5060 UDP)? > > Thanks a lot. > > Jason Lee > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090610/bac7391e/attachment-0002.html From mgg at giagnocavo.net Wed Jun 10 21:04:26 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 11 Jun 2009 00:04:26 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> How are you handling your FS box crashing? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Dalgliesh Sent: Wednesday, June 10, 2009 9:04 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Live Upgrade Techniques Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Wed Jun 10 21:41:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 11 Jun 2009 00:41:48 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> Message-ID: <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> By reporting it on Jira so it doesn't crash anymore :D On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote: > How are you handling your FS box crashing? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of John Dalgliesh > Sent: Wednesday, June 10, 2009 9:04 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Live Upgrade Techniques > > > Hi, > > I am slowly gaining confidence using FreeSWITCH in production, but > there > is one issue that I'm still wondering about: how are people upgrading > their FreeSWITCH installation binaries without dropping all current > calls? > > So far I have been upgrading in the dead of night, after pausing for 5 > minutes then dropping the stragglers, but this is hardly ideal. > > What I would like to do is to run an upgraded instance of FreeSWITCH > on > the same machine, and have it handle all new call packets, whereas > the old > instance continues to handle the existing call packets, until there > are no > more old calls left. > > I can think of about seven ways to accomplish this, but before I > dive into > the code I thought I'd better ask what everyone else has been doing :) > > (The only standard way I can think of doing this is to have a SIP > proxy > sitting in front of FS the whole time, just to handle these upgrade > windows. It seems like a bit of a waste.) > > So how are you handling your FS software upgrades? > > {P^/ > John > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Wed Jun 10 22:56:52 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 11 Jun 2009 01:56:52 -0400 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A30547D.4090101@gmail.com> References: <4A30547D.4090101@gmail.com> Message-ID: <4A309CA4.10609@freeswitch.org> jcromes at gmail.com wrote: > Thoughts anyone? > You can't please all of the people all of the time -Ray From diego.viola at gmail.com Wed Jun 10 23:17:15 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 11 Jun 2009 02:17:15 -0400 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A30547D.4090101@gmail.com> References: <4A30547D.4090101@gmail.com> Message-ID: <86a32abc0906102317i7a441a82s877762652b7d92f4@mail.gmail.com> I don't have anything against the web site, I like it, but I kinda agree that these banners are too big and a bit negative/unprofessional. Take this for example: http://www.freeswitch.org/ads/ad8.jpg When someone looks at that the first thing they get is a negative feeling and that's not good, doesn't matter what the message is, the negative feeling is there. So yeah, I kinda agree with you. Regards, Diego On Wed, Jun 10, 2009 at 8:49 PM, wrote: > My company is currently investigating a couple of projects that may take > me in the direction of FreeSwitch... In general, our management does > not often consider open source software for projects such as this, but > I've been successful in proving to them recently that open source can > deliver. > > FreeSwitch is a *very* professional and polished product, I can tell - > from the code and from the community. > > Unfortunately, I've been hesitant to send people to your webpage lately > because it went downhill a few weeks ago. Whenever I think about one of > our executives going to your webpage (after my recommendation) and > seeing a picture of people clanking beer glasses, or some idiot tied up > in phone cables, I cringe. I know you're advertising for ClueCon, but > honestly, some of those huge images on your front page really knock your > product down a peg in professionalism. > > Anyway, I'm pretty new to the community and I don't claim to be a web > designer. You have an excellent piece of software, but if I didn't > already know that about FreeSwitch, your webpage would not make a good > first impression. > > Please take that for what it's worth... I wanted to voice my opinion > because if I'm thinking it, others may be as well. > Thoughts anyone? > > J > > (Have I mentioned how awesome your source browser is though??!!) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/74e7bd07/attachment-0002.html From michal.bielicki at halo2.pl Thu Jun 11 03:47:46 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Thu, 11 Jun 2009 12:47:46 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: Message-ID: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> Am 11.06.2009 um 05:04 schrieb John Dalgliesh: > > Hi, > > I am slowly gaining confidence using FreeSWITCH in production, but > there > is one issue that I'm still wondering about: how are people upgrading > their FreeSWITCH installation binaries without dropping all current > calls? > > So far I have been upgrading in the dead of night, after pausing for 5 > minutes then dropping the stragglers, but this is hardly ideal. > > What I would like to do is to run an upgraded instance of FreeSWITCH > on > the same machine, and have it handle all new call packets, whereas > the old > instance continues to handle the existing call packets, until there > are no > more old calls left. > > I can think of about seven ways to accomplish this, but before I > dive into > the code I thought I'd better ask what everyone else has been doing :) > > (The only standard way I can think of doing this is to have a SIP > proxy > sitting in front of FS the whole time, just to handle these upgrade > windows. It seems like a bit of a waste.) > > So how are you handling your FS software upgrades? > > {P^/ > John > > We use freeswitch on solaris and just upgrade it to a new zfs which gets remounted to the old place and freeswitch gracefully restartet. On failure we can allways do a rollback, which takes between 2 and 10 seconds, so the dwntime is pretty acceptable. Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2453 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/93ef242c/attachment-0002.bin From yudha2008 at gmail.com Thu Jun 11 05:13:15 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 11 Jun 2009 17:43:15 +0530 Subject: [Freeswitch-users] FIFO through javascript Message-ID: Hi, I have configured inbound in FS SVN Trunk. i have written small program for inbound call to bridge. i have used session fifo session.execute( "fifo", "sales_fifo_1 out wait undef '/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" ); session.execute("bridge", "sofia/internal/1003%XXXXXXXXXXXX"); Inbound call pass through JavaScript session and play the voice file but it did not bridge to the extension 1003. It keep on playing the same voice file. how can i bridge the call after session.execute. session.execute( "fifo", "sales_fifo_1 out wait undef '/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" ); Can any one assist me to resolve the above problem -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8367970b/attachment-0002.html From brian at freeswitch.org Thu Jun 11 05:21:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 07:21:11 -0500 Subject: [Freeswitch-users] FIFO through javascript In-Reply-To: References: Message-ID: <949646D7-10E9-415F-B908-92AFA5DBF576@freeswitch.org> You can't... once you execute fifo your script has stopped. I think you have the idea that your script will keep running after you enter the fifo... /b On Jun 11, 2009, at 7:13 AM, Baskar wrote: > > Can any one assist me to resolve the above problem From larclap at yahoo.com Thu Jun 11 06:49:09 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 06:49:09 -0700 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log Message-ID: <003901c9ea9b$659eb4a0$30dc1de0$@com> In a dialplan, the action sets effective_caller_id_number to a value, however, in INFO, the displayed value is not the same as the set. Why? http://pastebin.freeswitch.org/9361 Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/e76fd6ca/attachment-0002.html From klaus.teller at gmx.net Thu Jun 11 07:19:47 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 11 Jun 2009 16:19:47 +0200 Subject: [Freeswitch-users] Rejecting calls without answering Message-ID: <20090611141947.246920@gmx.net> Hi Team, I'm still in need of a way to reject a call without answering it. I very much appreciate your help. Klaus. -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 From intralanman at freeswitch.org Thu Jun 11 07:23:22 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 11 Jun 2009 10:23:22 -0400 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <20090611141947.246920@gmx.net> References: <20090611141947.246920@gmx.net> Message-ID: <4A31135A.109@freeswitch.org> Klaus Teller wrote: > Hi Team, > > I'm still in need of a way to reject a call without answering it. I very much appreciate your help. > > Klaus. > -Ray From brian at freeswitch.org Thu Jun 11 07:24:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 09:24:06 -0500 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <003901c9ea9b$659eb4a0$30dc1de0$@com> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> Message-ID: <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> Are you doing an originate or a bridge? /b On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote: > In a dialplan, the action sets effective_caller_id_number to a > value, however, in INFO, the displayed value is not the same as the > set. Why? > > http://pastebin.freeswitch.org/9361 > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8887882c/attachment-0002.html From brian at freeswitch.org Thu Jun 11 07:24:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 09:24:30 -0500 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <20090611141947.246920@gmx.net> References: <20090611141947.246920@gmx.net> Message-ID: <44BA8761-1819-4B1C-A7BD-4178A37BEFCC@freeswitch.org> respond will do exactly that... try just hangup /b On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: > Hi Team, > > I'm still in need of a way to reject a call without answering it. I > very much appreciate your help. > > Klaus. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/29eb17d2/attachment-0002.html From larclap at yahoo.com Thu Jun 11 07:54:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 07:54:45 -0700 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> Message-ID: <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> Bridge From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 7:24 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log Are you doing an originate or a bridge? /b On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote: In a dialplan, the action sets effective_caller_id_number to a value, however, in INFO, the displayed value is not the same as the set. Why? http://pastebin.freeswitch.org/9361 Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/41380844/attachment-0002.html From brian at freeswitch.org Thu Jun 11 08:06:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 10:06:13 -0500 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> Message-ID: <9D918021-7578-4A1F-9C4C-BFF2485B3703@freeswitch.org> make sure you set it before the bridge. /b On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote: > Bridge > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Thursday, June 11, 2009 7:24 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log > > Are you doing an originate or a bridge? > > /b Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/639419f3/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 11 08:17:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 10:17:35 -0500 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A30547D.4090101@gmail.com> References: <4A30547D.4090101@gmail.com> Message-ID: <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> One important thing is that if we go around following everything everybody else says we become a follower in our field. I have had numerous people tell me what to do in the code, what to name things, what to eat for breakfast. Plain and simple, I will choose what to put on our website, when to put it there and what it says. You are welcome to your own opinion. I have no problem with it. If you say something I like we may even listen. Feel free to comment on anything else you find when browsing our community. BUT, If you have no sense of humor, you will not make it far in the open source telecom industry. If you want a more professional looking site, we do have some guys in suits on our FreeSWITCH Solutions site. http://www.freeswitchsolutions.com/ On Wed, Jun 10, 2009 at 7:49 PM, wrote: > My company is currently investigating a couple of projects that may take > me in the direction of FreeSwitch... In general, our management does > not often consider open source software for projects such as this, but > I've been successful in proving to them recently that open source can > deliver. > > FreeSwitch is a *very* professional and polished product, I can tell - > from the code and from the community. > > Unfortunately, I've been hesitant to send people to your webpage lately > because it went downhill a few weeks ago. Whenever I think about one of > our executives going to your webpage (after my recommendation) and > seeing a picture of people clanking beer glasses, or some idiot tied up > in phone cables, I cringe. I know you're advertising for ClueCon, but > honestly, some of those huge images on your front page really knock your > product down a peg in professionalism. > > Anyway, I'm pretty new to the community and I don't claim to be a web > designer. You have an excellent piece of software, but if I didn't > already know that about FreeSwitch, your webpage would not make a good > first impression. > > Please take that for what it's worth... I wanted to voice my opinion > because if I'm thinking it, others may be as well. > Thoughts anyone? > > J > > (Have I mentioned how awesome your source browser is though??!!) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/18121d58/attachment-0002.html From larclap at yahoo.com Thu Jun 11 08:26:45 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 08:26:45 -0700 Subject: [Freeswitch-users] Help understanding DEBUG and INFO log In-Reply-To: <9D918021-7578-4A1F-9C4C-BFF2485B3703@freeswitch.org> References: <003901c9ea9b$659eb4a0$30dc1de0$@com> <4D9544D6-5BF3-44BE-A8C3-8751F64D6C0F@freeswitch.org> <005f01c9eaa4$8f84c8f0$ae8e5ad0$@com> <9D918021-7578-4A1F-9C4C-BFF2485B3703@freeswitch.org> Message-ID: <007e01c9eaa9$07e8ce00$17ba6a00$@com> It was. You can see the set at line 2 was done before the bridge at line 4. What am I missing? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 8:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log make sure you set it before the bridge. /b On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote: Bridge From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 7:24 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log Are you doing an originate or a bridge? /b Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/6d428e66/attachment-0002.html From steveu at coppice.org Thu Jun 11 09:09:06 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 12 Jun 2009 00:09:06 +0800 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> Message-ID: <4A312C22.1010504@coppice.org> Anthony Minessale wrote: > One important thing is that if we go around following everything > everybody else says > we become a follower in our field. > > I have had numerous people tell me what to do in the code, what to > name things, what to eat for breakfast. > Plain and simple, I will choose what to put on our website, when to > put it there and what it says. > > You are welcome to your own opinion. I have no problem with it. If > you say something I like we may > even listen. Feel free to comment on anything else you find when > browsing our community. > > BUT, > > If you have no sense of humor, you will not make it far in the open > source telecom industry. > > If you want a more professional looking site, we do have some guys in > suits on our FreeSWITCH Solutions site. > http://www.freeswitchsolutions.com/ The main reason www.freeswitchsolutions.com looks more professional that www.freeswitch.org is not the content of the pictures but their size. The pictures at the top of the www.freeswitch.org are too big and in your face. They completely dominate the screen when it appears. Steve From klaus.teller at gmx.net Thu Jun 11 09:21:30 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 11 Jun 2009 18:21:30 +0200 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <44BA8761-1819-4B1C-A7BD-4178A37BEFCC@freeswitch.org> References: <20090611141947.246920@gmx.net> <44BA8761-1819-4B1C-A7BD-4178A37BEFCC@freeswitch.org> Message-ID: <20090611162130.284160@gmx.net> Hi Folks, Here is what i'm observing. When i connect with Xlite (registered device) and call the 9444 extension (see below), Freeswitch does hangup as i would like it to. But when i call via gafachi, something weird happens. What i can see is that Freeswitch sends a hangup signal (service temporarily not available) to Gafachi, but the guys keep sending back the very same call. It looks to me like a Gafachi issue. But can anything else be done on the Freeswitch side? I'm attaching the logs for the gafachi call this. All you see in there is just one single call. You will see that a new channel is created more than once. Any thought? Klaus. The gafachi respond extension (under conf/dialplan/public/reject.xml): The gafachi profile (under conf/sip_profiles/external/gafachi.xml): The Xlite respond test extension (in default.xml): Any idea? > respond will do exactly that... try just hangup > > /b > > On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: > > > Hi Team, > > > > I'm still in need of a way to reject a call without answering it. I > > very much appreciate your help. > > > > Klaus. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 -------------- next part -------------- 2009-06-11 12:14:34.870820 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [6867d2a1-e26a-4cab-9122-7ecab2b9397f] 2009-06-11 12:14:34.870820 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:34.870820 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 273544867 273544867 IN IP4 67.216.37.18 s=session c=IN IP4 67.216.37.18 t=0 0 m=audio 35116 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:34.870820 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:34.870820 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:34.870820 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_NEW 2009-06-11 12:14:34.874889 [DEBUG] switch_core_state_machine.c:403 (sofia/external/6473671811 at sip.gafachi.com) State NEW 2009-06-11 12:14:34.874889 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:34.874889 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:34.874889 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:34.878827 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:34.878827 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:34.878827 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:34.882811 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:34.882811 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:34.882811 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.882811 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-11 12:14:34.882811 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:34.882811 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:34.882811 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:34.882811 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_CLEARING 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_CLEARING 2009-06-11 12:14:34.882811 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_CLEARING 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:34.886819 [DEBUG] switch_core_session.c:1067 Session 1 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:34.886819 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:34.886819 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:34.886819 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:34.886819 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:38.559058 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [f1544b7b-0176-447d-88f0-db6a422fb89e] 2009-06-11 12:14:38.559058 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:38.559058 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 1686340962 1686340962 IN IP4 67.216.37.18 s=session c=IN IP4 67.216.37.18 t=0 0 m=audio 43928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:38.559058 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:38.559058 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:38.559058 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:38.559058 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:38.559058 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:38.559058 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:38.559058 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:38.559058 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:38.559058 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:38.559058 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:38.559058 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:38.563057 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:38.563057 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:38.563057 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:38.563057 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:38.563057 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:38.563057 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:38.567118 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:38.567118 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:38.567118 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:38.567118 [DEBUG] switch_core_session.c:1067 Session 2 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:38.567118 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:38.567118 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:38.567118 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:38.567118 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:42.471295 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [dcf89ba0-7932-4cc8-a337-5f807ae3c394] 2009-06-11 12:14:42.471295 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:42.471295 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 1254501935 1254501935 IN IP4 67.216.37.17 s=session c=IN IP4 67.216.37.17 t=0 0 m=audio 34272 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:42.471295 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:42.471295 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:42.471295 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:42.471295 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:42.471295 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.471295 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:42.475309 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:42.475309 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:42.475309 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:42.475309 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:42.479287 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:42.479287 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:42.479287 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:42.479287 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:42.479287 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:42.479287 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.479287 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:42.479287 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:42.479287 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:42.479287 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:42.483297 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:42.483297 [DEBUG] switch_core_session.c:1067 Session 3 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:42.483297 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:42.483297 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:42.483297 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:42.483297 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:44.147389 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [7e608276-0901-4525-8822-7d4c07283466] 2009-06-11 12:14:44.147389 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:44.147389 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 36880947 36880947 IN IP4 67.216.37.18 s=session c=IN IP4 67.216.37.18 t=0 0 m=audio 59904 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:44.147389 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:44.147389 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:44.151433 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:44.151433 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:44.151433 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:44.151433 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:44.151433 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:44.151433 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:44.155404 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:44.155404 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:44.155404 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:44.155404 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:44.155404 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:44.155404 [DEBUG] switch_core_session.c:1067 Session 4 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:44.155404 [NOTICE] switch_core_session.c:1085 Session 4 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:44.155404 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:44.155404 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:44.155404 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep 2009-06-11 12:14:46.131565 [NOTICE] switch_channel.c:602 New Channel sofia/external/6473671811 at sip.gafachi.com [0e736a20-8f6b-4f0a-941a-fe00e277bb80] 2009-06-11 12:14:46.135599 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [received][100] 2009-06-11 12:14:46.135599 [DEBUG] sofia.c:3107 Remote SDP: v=0 o=root 1258094781 1258094781 IN IP4 67.216.37.17 s=session c=IN IP4 67.216.37.17 t=0 0 m=audio 16632 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 2009-06-11 12:14:46.135599 [DEBUG] sofia_glue.c:3059 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-06-11 12:14:46.135599 [DEBUG] sofia_glue.c:2017 Set Codec sofia/external/6473671811 at sip.gafachi.com PCMU/8000 20 ms 160 samples 2009-06-11 12:14:46.135599 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_NEW 2009-06-11 12:14:46.135599 [DEBUG] switch_core_state_machine.c:403 (sofia/external/6473671811 at sip.gafachi.com) State NEW 2009-06-11 12:14:46.135599 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 101 2009-06-11 12:14:46.135599 [DEBUG] sofia.c:3266 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_NEW -> CS_INIT 2009-06-11 12:14:46.135599 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_INIT 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:83 sofia/external/6473671811 at sip.gafachi.com SOFIA INIT 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:111 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_INIT -> CS_ROUTING 2009-06-11 12:14:46.139548 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:480 (sofia/external/6473671811 at sip.gafachi.com) State INIT going to sleep 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_ROUTING 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:130 sofia/external/6473671811 at sip.gafachi.com SOFIA ROUTING 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:78 sofia/external/6473671811 at sip.gafachi.com Standard ROUTING 2009-06-11 12:14:46.139548 [INFO] mod_dialplan_xml.c:252 Processing unknown->18664591152 in context public Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->unloop] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->outside_call] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Absolute Condition [outside_call] Dialplan: sofia/external/6473671811 at sip.gafachi.com Action set(outside_call=true) Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->call_debug] continue=true Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_extensions] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_extensions] destination_number(18664591152) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->public_did] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (FAIL) [public_did] destination_number(18664591152) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com parsing [public->gafachi-extension] continue=false Dialplan: sofia/external/6473671811 at sip.gafachi.com Regex (PASS) [gafachi-extension] destination_number(18664591152) =~ /^1866.*$/ break=on-false Dialplan: sofia/external/6473671811 at sip.gafachi.com Action respond(503) 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:114 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_ROUTING -> CS_EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:483 (sofia/external/6473671811 at sip.gafachi.com) State ROUTING going to sleep 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] mod_sofia.c:173 sofia/external/6473671811 at sip.gafachi.com SOFIA EXECUTE 2009-06-11 12:14:46.139548 [DEBUG] switch_core_state_machine.c:151 sofia/external/6473671811 at sip.gafachi.com Standard EXECUTE EXECUTE sofia/external/6473671811 at sip.gafachi.com set(outside_call=true) 2009-06-11 12:14:46.139548 [DEBUG] mod_dptools.c:748 sofia/external/6473671811 at sip.gafachi.com SET [outside_call]=[true] EXECUTE sofia/external/6473671811 at sip.gafachi.com respond(503) 2009-06-11 12:14:46.147562 [DEBUG] mod_dptools.c:710 sofia/external/6473671811 at sip.gafachi.com receive message [RESPOND] 2009-06-11 12:14:46.147562 [DEBUG] mod_sofia.c:1406 Responding with 503 [Service Unavailable] 2009-06-11 12:14:46.147562 [DEBUG] switch_core_session.c:630 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.147562 [DEBUG] sofia.c:3100 Channel sofia/external/6473671811 at sip.gafachi.com entering state [terminated][503] 2009-06-11 12:14:46.147562 [NOTICE] sofia.c:3660 Hangup sofia/external/6473671811 at sip.gafachi.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-06-11 12:14:46.147562 [DEBUG] switch_core_state_machine.c:490 (sofia/external/6473671811 at sip.gafachi.com) State EXECUTE going to sleep 2009-06-11 12:14:46.147562 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_HANGUP 2009-06-11 12:14:46.147562 [DEBUG] switch_channel.c:1683 Send signal sofia/external/6473671811 at sip.gafachi.com [KILL] 2009-06-11 12:14:46.147562 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP 2009-06-11 12:14:46.152178 [DEBUG] mod_sofia.c:306 sofia/external/6473671811 at sip.gafachi.com Overriding SIP cause 503 with 503 from the other leg 2009-06-11 12:14:46.152178 [DEBUG] mod_sofia.c:338 Channel sofia/external/6473671811 at sip.gafachi.com hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:46 sofia/external/6473671811 at sip.gafachi.com Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:433 (sofia/external/6473671811 at sip.gafachi.com) State HANGUP going to sleep 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:475 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_HANGUP -> CS_REPORTING 2009-06-11 12:14:46.152178 [DEBUG] switch_core_session.c:933 Send signal sofia/external/6473671811 at sip.gafachi.com [BREAK] 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:397 (sofia/external/6473671811 at sip.gafachi.com) Running State Change CS_REPORTING 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:53 sofia/external/6473671811 at sip.gafachi.com Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:607 (sofia/external/6473671811 at sip.gafachi.com) State REPORTING going to sleep 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6473671811 at sip.gafachi.com) State Change CS_REPORTING -> CS_DESTROY 2009-06-11 12:14:46.152178 [DEBUG] switch_core_session.c:1067 Session 5 (sofia/external/6473671811 at sip.gafachi.com) Locked, Waiting on external entities 2009-06-11 12:14:46.152178 [NOTICE] switch_core_session.c:1085 Session 5 (sofia/external/6473671811 at sip.gafachi.com) Ended 2009-06-11 12:14:46.152178 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/6473671811 at sip.gafachi.com [CS_DESTROY] 2009-06-11 12:14:46.152178 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY 2009-06-11 12:14:46.152178 [DEBUG] mod_sofia.c:255 sofia/external/6473671811 at sip.gafachi.com SOFIA DESTROY 2009-06-11 12:14:46.157559 [DEBUG] switch_core_state_machine.c:60 sofia/external/6473671811 at sip.gafachi.com Standard DESTROY 2009-06-11 12:14:46.157559 [DEBUG] switch_core_state_machine.c:559 (sofia/external/6473671811 at sip.gafachi.com) State DESTROY going to sleep From krice at freeswitch.org Thu Jun 11 09:31:13 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 11 Jun 2009 11:31:13 -0500 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: <20090611162130.284160@gmx.net> Message-ID: Hah they are just retrying the call to see if they get a different answer the 2nd and 3rd time around... This is common unfortunately since 503 in the VoIP world is typically interpreted by the PSTN world as a "Temp congestion" (and rightly so) so they will retry and not fail the call... You can try responding w/ 486 Busy if you know the call doesn't need to fail somewhere else... > From: Klaus Teller > Reply-To: > Date: Thu, 11 Jun 2009 18:21:30 +0200 > To: > Subject: Re: [Freeswitch-users] Rejecting calls without answering > > Hi Folks, > > Here is what i'm observing. When i connect with Xlite (registered device) and > call the 9444 extension (see below), Freeswitch does hangup as i would like it > to. > > But when i call via gafachi, something weird happens. What i can see is that > Freeswitch sends a hangup signal (service temporarily not available) to > Gafachi, but the guys keep sending back the very same call. > > > It looks to me like a Gafachi issue. But can anything else be done on the > Freeswitch side? > > I'm attaching the logs for the gafachi call this. All you see in there is just > one single call. You will see that a new channel is created more than once. > > Any thought? > > Klaus. > > The gafachi respond extension (under conf/dialplan/public/reject.xml): > > > > > > > > > > The gafachi profile (under conf/sip_profiles/external/gafachi.xml): > > > > > > > > > The Xlite respond test extension (in default.xml): > > > > > > > > Any idea? > > > > > > >> respond will do exactly that... try just hangup >> >> /b >> >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: >> >>> Hi Team, >>> >>> I'm still in need of a way to reject a call without answering it. I >>> very much appreciate your help. >>> >>> Klaus. >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> > > -- > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss > f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jun 11 09:36:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 11:36:47 -0500 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A312C22.1010504@coppice.org> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> <4A312C22.1010504@coppice.org> Message-ID: <191c3a030906110936i1ad40654y4635c5532d46150@mail.gmail.com> Ok, let's try them at half size. On Thu, Jun 11, 2009 at 11:09 AM, Steve Underwood wrote: > Anthony Minessale wrote: > > One important thing is that if we go around following everything > > everybody else says > > we become a follower in our field. > > > > I have had numerous people tell me what to do in the code, what to > > name things, what to eat for breakfast. > > Plain and simple, I will choose what to put on our website, when to > > put it there and what it says. > > > > You are welcome to your own opinion. I have no problem with it. If > > you say something I like we may > > even listen. Feel free to comment on anything else you find when > > browsing our community. > > > > BUT, > > > > If you have no sense of humor, you will not make it far in the open > > source telecom industry. > > > > If you want a more professional looking site, we do have some guys in > > suits on our FreeSWITCH Solutions site. > > http://www.freeswitchsolutions.com/ > The main reason www.freeswitchsolutions.com > looks more professional that > www.freeswitch.org is not the content of the pictures but their size. > The pictures at the top of the www.freeswitch.org are too big and in > your face. They completely dominate the screen when it appears. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/2f8168d0/attachment-0002.html From klaus.teller at gmx.net Thu Jun 11 09:40:24 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 11 Jun 2009 18:40:24 +0200 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: References: Message-ID: <20090611164024.22470@gmx.net> Excellent! Thank you everybody. Response 486 did the trick. Klaus. -------- Original-Nachricht -------- > Datum: Thu, 11 Jun 2009 11:31:13 -0500 > Von: Ken Rice > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Rejecting calls without answering > Hah they are just retrying the call to see if they get a different answer > the 2nd and 3rd time around... This is common unfortunately since 503 in > the > VoIP world is typically interpreted by the PSTN world as a "Temp > congestion" > (and rightly so) so they will retry and not fail the call... You can try > responding w/ 486 Busy if you know the call doesn't need to fail somewhere > else... > > > > From: Klaus Teller > > Reply-To: > > Date: Thu, 11 Jun 2009 18:21:30 +0200 > > To: > > Subject: Re: [Freeswitch-users] Rejecting calls without answering > > > > Hi Folks, > > > > Here is what i'm observing. When i connect with Xlite (registered > device) and > > call the 9444 extension (see below), Freeswitch does hangup as i would > like it > > to. > > > > But when i call via gafachi, something weird happens. What i can see is > that > > Freeswitch sends a hangup signal (service temporarily not available) to > > Gafachi, but the guys keep sending back the very same call. > > > > > > It looks to me like a Gafachi issue. But can anything else be done on > the > > Freeswitch side? > > > > I'm attaching the logs for the gafachi call this. All you see in there > is just > > one single call. You will see that a new channel is created more than > once. > > > > Any thought? > > > > Klaus. > > > > The gafachi respond extension (under conf/dialplan/public/reject.xml): > > > > > > expression="^866.*$"> > > > > > > > > > > > > > > The gafachi profile (under conf/sip_profiles/external/gafachi.xml): > > > > > > > > > > > > > > > > > > The Xlite respond test extension (in default.xml): > > > > > > > > > > > > > > > > Any idea? > > > > > > > > > > > > > >> respond will do exactly that... try just hangup > >> > >> /b > >> > >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: > >> > >>> Hi Team, > >>> > >>> I'm still in need of a way to reject a call without answering it. I > >>> very much appreciate your help. > >>> > >>> Klaus. > >> > >> Brian West > >> brian at freeswitch.org > >> > >> -- Meet us at ClueCon! http://www.cluecon.com > >> > >> > >> > >> > > > > -- > > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und > Telefonanschluss > > f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From diego.viola at gmail.com Thu Jun 11 09:45:56 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 11 Jun 2009 12:45:56 -0400 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A312C22.1010504@coppice.org> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> <4A312C22.1010504@coppice.org> Message-ID: <86a32abc0906110945w27829e4ei1cff1cbfe3409bac@mail.gmail.com> That's what I tried to say, I didn't expressed myself well, sorry. On Thu, Jun 11, 2009 at 12:09 PM, Steve Underwood wrote: > Anthony Minessale wrote: > > One important thing is that if we go around following everything > > everybody else says > > we become a follower in our field. > > > > I have had numerous people tell me what to do in the code, what to > > name things, what to eat for breakfast. > > Plain and simple, I will choose what to put on our website, when to > > put it there and what it says. > > > > You are welcome to your own opinion. I have no problem with it. If > > you say something I like we may > > even listen. Feel free to comment on anything else you find when > > browsing our community. > > > > BUT, > > > > If you have no sense of humor, you will not make it far in the open > > source telecom industry. > > > > If you want a more professional looking site, we do have some guys in > > suits on our FreeSWITCH Solutions site. > > http://www.freeswitchsolutions.com/ > The main reason www.freeswitchsolutions.com > looks more professional that > www.freeswitch.org is not the content of the pictures but their size. > The pictures at the top of the www.freeswitch.org are too big and in > your face. They completely dominate the screen when it appears. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/fc025a16/attachment-0002.html From freeswitch-users-list at metik.com Thu Jun 11 09:53:28 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 11 Jun 2009 12:53:28 -0400 Subject: [Freeswitch-users] Rejecting calls without answering In-Reply-To: References: Message-ID: Although a little overkill, RFC3398 also describes some desirable interop behavior between ISUP, ISDN and SIP. (From "7.2.4.1 ISDN Cause Code to Status Code Mapping") [...] ISUP Cause value SIP response ---------------- ------------ 1 unallocated number 404 Not Found 2 no route to network 404 Not found 3 no route to destination 404 Not found 16 normal call clearing --- (*) 17 user busy 486 Busy here 18 no user responding 408 Request Timeout 19 no answer from the user 480 Temporarily unavailable 20 subscriber absent 480 Temporarily unavailable 21 call rejected 403 Forbidden (+) 22 number changed (w/o diagnostic) 410 Gone 22 number changed (w/ diagnostic) 301 Moved Permanently 23 redirection to new destination 410 Gone 26 non-selected user clearing 404 Not Found (=) 27 destination out of order 502 Bad Gateway 28 address incomplete 484 Address incomplete 29 facility rejected 501 Not implemented 31 normal unspecified 480 Temporarily unavailable [...] Resource unavailable This kind of cause value indicates a temporary failure. A 'Retry-After' header MAY be added to the response if appropriate. ISUP Cause value SIP response ---------------- ------------ 34 no circuit available 503 Service unavailable 38 network out of order 503 Service unavailable 41 temporary failure 503 Service unavailable 42 switching equipment congestion 503 Service unavailable 47 resource unavailable 503 Service unavailable ----- Original Message ----- From: "Ken Rice" To: Sent: Thursday, June 11, 2009 12:31 PM Subject: Re: [Freeswitch-users] Rejecting calls without answering Hah they are just retrying the call to see if they get a different answer the 2nd and 3rd time around... This is common unfortunately since 503 in the VoIP world is typically interpreted by the PSTN world as a "Temp congestion" (and rightly so) so they will retry and not fail the call... You can try responding w/ 486 Busy if you know the call doesn't need to fail somewhere else... > From: Klaus Teller > Reply-To: > Date: Thu, 11 Jun 2009 18:21:30 +0200 > To: > Subject: Re: [Freeswitch-users] Rejecting calls without answering > > Hi Folks, > > Here is what i'm observing. When i connect with Xlite (registered device) > and > call the 9444 extension (see below), Freeswitch does hangup as i would > like it > to. > > But when i call via gafachi, something weird happens. What i can see is > that > Freeswitch sends a hangup signal (service temporarily not available) to > Gafachi, but the guys keep sending back the very same call. > > > It looks to me like a Gafachi issue. But can anything else be done on the > Freeswitch side? > > I'm attaching the logs for the gafachi call this. All you see in there is > just > one single call. You will see that a new channel is created more than > once. > > Any thought? > > Klaus. > > The gafachi respond extension (under conf/dialplan/public/reject.xml): > > > > > > > > > > The gafachi profile (under conf/sip_profiles/external/gafachi.xml): > > > > > > > > > The Xlite respond test extension (in default.xml): > > > > > > > > Any idea? > > > > > > >> respond will do exactly that... try just hangup >> >> /b >> >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote: >> >>> Hi Team, >>> >>> I'm still in need of a way to reject a call without answering it. I >>> very much appreciate your help. >>> >>> Klaus. >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> > > -- > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss > f?r nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jun 11 09:54:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 11:54:32 -0500 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> Message-ID: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> or you can put a sip proxy in front of 2 boxes where you can control the flow of traffic. when you want to upgrade one, take all the traffic off of it by forcing all calls to the other box, upgrade it then shift the traffic to the new one. if that goes well, upgrade the other one too. On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki wrote: > > Am 11.06.2009 um 05:04 schrieb John Dalgliesh: > > >> Hi, >> >> I am slowly gaining confidence using FreeSWITCH in production, but there >> is one issue that I'm still wondering about: how are people upgrading >> their FreeSWITCH installation binaries without dropping all current calls? >> >> So far I have been upgrading in the dead of night, after pausing for 5 >> minutes then dropping the stragglers, but this is hardly ideal. >> >> What I would like to do is to run an upgraded instance of FreeSWITCH on >> the same machine, and have it handle all new call packets, whereas the old >> instance continues to handle the existing call packets, until there are no >> more old calls left. >> >> I can think of about seven ways to accomplish this, but before I dive into >> the code I thought I'd better ask what everyone else has been doing :) >> >> (The only standard way I can think of doing this is to have a SIP proxy >> sitting in front of FS the whole time, just to handle these upgrade >> windows. It seems like a bit of a waste.) >> >> So how are you handling your FS software upgrades? >> >> {P^/ >> John >> >> >> > > We use freeswitch on solaris and just upgrade it to a new zfs which gets > remounted to the old place and freeswitch gracefully restartet. On failure > we can allways do a rollback, which takes between 2 and 10 seconds, so the > dwntime is pretty acceptable. > > Michal Bielicki > Leiter der Niederlassung > HaloKwadrat Sp. z o.o. > Niederlassung Kleinmachnow > Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P > Ust.Id.: DE261885536 > Geschaeftsfuehrer: Aleksander Wiercinski > Meiereifeld 2b, 14532 Kleinmachnow > t. +49 33203 263220 > f. +49 33203 263229 sip. info at halokwadrat.de > e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de > Hauptgesch?ftsstelle: > Halo Kwadrat Sp. z o.o. > ul. Polna 46/14 > 00-644 Warszawa, Polen > EIngetragen im HRB Warszawa, KRS 0000153539 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/dda20ded/attachment-0002.html From mgg at giagnocavo.net Thu Jun 11 10:33:53 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 11 Jun 2009 13:33:53 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> Exactly. You probably want to have something like this anyways, so that when someone accidentally unplugs the system, or the disks/CPU/RAM crash, you're not stuck. That is, until FreeSWITCH can record its internal state to some inter-machine memory so we can have hot failover. ;) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, June 11, 2009 10:55 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques or you can put a sip proxy in front of 2 boxes where you can control the flow of traffic. when you want to upgrade one, take all the traffic off of it by forcing all calls to the other box, upgrade it then shift the traffic to the new one. if that goes well, upgrade the other one too. On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki wrote: Am 11.06.2009 um 05:04 schrieb John Dalgliesh: Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John We use freeswitch on solaris and just upgrade it to a new zfs which gets remounted to the old place and freeswitch gracefully restartet. On failure we can allways do a rollback, which takes between 2 and 10 seconds, so the dwntime is pretty acceptable. Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c0762d9d/attachment-0002.html From msc at freeswitch.org Thu Jun 11 10:42:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 10:42:49 -0700 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> Message-ID: <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote: > Exactly. You probably want to have something like this anyways, so that > when someone accidentally unplugs the system, or the disks/CPU/RAM crash, > you?re not stuck. > > > > That is, until FreeSWITCH can record its internal state to some > inter-machine memory so we can have hot failover. ;) > > > I think that's going to be in 1.0.5. :) > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Thursday, June 11, 2009 10:55 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques > > > > or you can put a sip proxy in front of 2 boxes where you can control the > flow of traffic. > when you want to upgrade one, take all the traffic off of it by forcing all > calls to the other box, upgrade it then shift the traffic to the new one. > if that goes well, upgrade the other one too. > > > On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki > wrote: > > > Am 11.06.2009 um 05:04 schrieb John Dalgliesh: > > > > > Hi, > > I am slowly gaining confidence using FreeSWITCH in production, but there > is one issue that I'm still wondering about: how are people upgrading > their FreeSWITCH installation binaries without dropping all current calls? > > So far I have been upgrading in the dead of night, after pausing for 5 > minutes then dropping the stragglers, but this is hardly ideal. > > What I would like to do is to run an upgraded instance of FreeSWITCH on > the same machine, and have it handle all new call packets, whereas the old > instance continues to handle the existing call packets, until there are no > more old calls left. > > I can think of about seven ways to accomplish this, but before I dive into > the code I thought I'd better ask what everyone else has been doing :) > > (The only standard way I can think of doing this is to have a SIP proxy > sitting in front of FS the whole time, just to handle these upgrade > windows. It seems like a bit of a waste.) > > So how are you handling your FS software upgrades? > > {P^/ > John > > > > We use freeswitch on solaris and just upgrade it to a new zfs which gets > remounted to the old place and freeswitch gracefully restartet. On failure > we can allways do a rollback, which takes between 2 and 10 seconds, so the > dwntime is pretty acceptable. > > Michal Bielicki > Leiter der Niederlassung > HaloKwadrat Sp. z o.o. > Niederlassung Kleinmachnow > Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P > Ust.Id.: DE261885536 > Geschaeftsfuehrer: Aleksander Wiercinski > Meiereifeld 2b, 14532 Kleinmachnow > t. +49 33203 263220 > f. +49 33203 263229 sip. info at halokwadrat.de > e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de > Hauptgesch?ftsstelle: > Halo Kwadrat Sp. z o.o. > ul. Polna 46/14 > 00-644 Warszawa, Polen > EIngetragen im HRB Warszawa, KRS 0000153539 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/030512c7/attachment-0002.html From jcromes at gmail.com Thu Jun 11 10:43:40 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Thu, 11 Jun 2009 12:43:40 -0500 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> Message-ID: <4A31424C.4020501@gmail.com> Haha, good point on the FreeSwitch Solutions site... Suits - very professional. =) Please don't think I'm telling you guys what to do, I know you don't need that. It IS your software and your site, and you've done a HELL of a job with it so far. It was just a thought. Sorry if I offended. Anthony Minessale wrote: > One important thing is that if we go around following everything > everybody else says > we become a follower in our field. > > I have had numerous people tell me what to do in the code, what to > name things, what to eat for breakfast. > Plain and simple, I will choose what to put on our website, when to > put it there and what it says. > > You are welcome to your own opinion. I have no problem with it. If > you say something I like we may > even listen. Feel free to comment on anything else you find when > browsing our community. > > BUT, > > If you have no sense of humor, you will not make it far in the open > source telecom industry. > > If you want a more professional looking site, we do have some guys in > suits on our FreeSWITCH Solutions site. > http://www.freeswitchsolutions.com/ > From msc at freeswitch.org Thu Jun 11 10:47:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 10:47:24 -0700 Subject: [Freeswitch-users] Web page thoughts In-Reply-To: <4A31424C.4020501@gmail.com> References: <4A30547D.4090101@gmail.com> <191c3a030906110817q323edae7i13f6dc3b1f97dd8e@mail.gmail.com> <4A31424C.4020501@gmail.com> Message-ID: <87f2f3b90906111047v1febdbffsdc0b59528980b1e8@mail.gmail.com> On Thu, Jun 11, 2009 at 10:43 AM, wrote: > Haha, good point on the FreeSwitch Solutions site... Suits - very > professional. =) > Please don't think I'm telling you guys what to do, I know you don't > need that. It IS your software and your site, and you've done a HELL of > a job with it so far. > > It was just a thought. Sorry if I offended. > No offense taken. We DO appreciate feedback, unsolicited or otherwise, but we don't always agree with it. Definitely show your bosses the FSS site. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/500b8c81/attachment-0002.html From johnd at defyne.org Thu Jun 11 11:00:24 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 04:00:24 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> Message-ID: OK thanks that is what I thought the general way of doing it would be. But it seems a bit wasteful to have that SIP proxy there the whole time especially when I am using FS in the role of an SBC. The problem with the graceful restart of course is that you have to wait for the calls count to get to zero, which may never happen. It's 3:30am here in Sydney now and I just checked FS: 20 calls in progress still! So what I plan to do is add a '--upgrade' cmd line arg to FS. This will make the new instance contact the old one on a unix socket and receive a dup of its SIP socket fd(s) via a SCM_RIGHTS sendmsg. It will use those for sending and the unix socket for receiving. Meanwhile the old instance will pass any packets with unknown Call-Ids over the unix socket to the new instance, instead of handling them itself. When the old instance has no calls left, it shuts down. The new instance detects the unix socket is closed and switches to reading from the SIP socket (which would have buffered any unread packets - so nothing is lost). Sound good? I realise this will be 90% in libsofia but I've read teh code and it seems very do-able. Anyone interested in my changes will of course be most welcome to them. The runner-up approach I considered was to make a kernel module that extends iptables with a filter that can extract the Call-Id and look it up in a table that is somehow populated from FS. Maybe this exists already? Kind of a SIP proxy lite that can be enabled on the server machine when needed. Anyway that lost out as it's more work and even less portable. {P^/ John On Thu, 11 Jun 2009 at 11:54 -0500, Anthony Minessale wrote: > > or you can put a sip proxy in front of 2 boxes where you can control the > flow of traffic. > when you want to upgrade one, take all the traffic off of it by forcing all > calls to the other box, upgrade it then shift the traffic to the new one. > if that goes well, upgrade the other one too. > > > > On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki > wrote: > >> >> Am 11.06.2009 um 05:04 schrieb John Dalgliesh: >> >> >>> Hi, >>> >>> I am slowly gaining confidence using FreeSWITCH in production, but there >>> is one issue that I'm still wondering about: how are people upgrading >>> their FreeSWITCH installation binaries without dropping all current calls? >>> >>> So far I have been upgrading in the dead of night, after pausing for 5 >>> minutes then dropping the stragglers, but this is hardly ideal. >>> >>> What I would like to do is to run an upgraded instance of FreeSWITCH on >>> the same machine, and have it handle all new call packets, whereas the old >>> instance continues to handle the existing call packets, until there are no >>> more old calls left. >>> >>> I can think of about seven ways to accomplish this, but before I dive into >>> the code I thought I'd better ask what everyone else has been doing :) >>> >>> (The only standard way I can think of doing this is to have a SIP proxy >>> sitting in front of FS the whole time, just to handle these upgrade >>> windows. It seems like a bit of a waste.) >>> >>> So how are you handling your FS software upgrades? >>> >>> {P^/ >>> John >>> >>> >>> >> >> We use freeswitch on solaris and just upgrade it to a new zfs which gets >> remounted to the old place and freeswitch gracefully restartet. On failure >> we can allways do a rollback, which takes between 2 and 10 seconds, so the >> dwntime is pretty acceptable. >> >> Michal Bielicki >> Leiter der Niederlassung >> HaloKwadrat Sp. z o.o. >> Niederlassung Kleinmachnow >> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P >> Ust.Id.: DE261885536 >> Geschaeftsfuehrer: Aleksander Wiercinski >> Meiereifeld 2b, 14532 Kleinmachnow >> t. +49 33203 263220 >> f. +49 33203 263229 sip. info at halokwadrat.de >> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de >> Hauptgesch?ftsstelle: >> Halo Kwadrat Sp. z o.o. >> ul. Polna 46/14 >> 00-644 Warszawa, Polen >> EIngetragen im HRB Warszawa, KRS 0000153539 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > From lcm at marshap.com Thu Jun 11 11:04:07 2009 From: lcm at marshap.com (Larry Marshall) Date: Thu, 11 Jun 2009 11:04:07 -0700 Subject: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition Message-ID: <00d501c9eabf$03c7ab00$0b570100$@com> http://pastebin.freeswitch.org/9365 I do not know what I am doing wrong. I am trying to set the effective_caller_id_name and _number depending on the originating extension. I tried: and and But each got substituted with the name of the extension in the log: Dialplan: sofia/internal/1000 at 192.168.10.29 Regex (FAIL) [Long Distance - flowroute] () =~ /^100[09]$/ break=on-true where the extension looks like: Info from the log shows variable_sip_from_user: [1000] Caller-Caller-ID-Number: [1000] Can anyone help? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8be3a0b6/attachment-0002.html From johnd at defyne.org Thu Jun 11 11:13:39 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 04:13:39 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> Message-ID: I assume he's talking about hardware failures here :P But to answer the question: crashes are easy to deal with. With a crash you have lost the calls that are in progress anyway; you don't have to manage a gradual transition. Currently, since FS is quite quick to start up, I am just relaunching it immediately. But when I have a second box up and running what I'll do is just add the IP of the dead machine as another IP of the second box, and then it will take all the old machine's traffic. That is the plan anyway. I've seen some commercial boxes that use a similar trick. (I've only seen one crash that wasn't my fault. Something to do with terminating a bridge: when the first leg gets a hangup it hangs up the other leg on its own thread... which can cause problems if the other leg was doing something funky at the time. Leads to a heap corruption. Doesn't happen with MALLOC_CHECK_ set so I'm just leaving it set for now :) {P^/ On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote: > > By reporting it on Jira so it doesn't crash anymore :D > > > On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote: > >> How are you handling your FS box crashing? >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of John Dalgliesh >> Sent: Wednesday, June 10, 2009 9:04 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Live Upgrade Techniques >> >> >> Hi, >> >> I am slowly gaining confidence using FreeSWITCH in production, but >> there >> is one issue that I'm still wondering about: how are people upgrading >> their FreeSWITCH installation binaries without dropping all current >> calls? >> >> So far I have been upgrading in the dead of night, after pausing for 5 >> minutes then dropping the stragglers, but this is hardly ideal. >> >> What I would like to do is to run an upgraded instance of FreeSWITCH >> on >> the same machine, and have it handle all new call packets, whereas >> the old >> instance continues to handle the existing call packets, until there >> are no >> more old calls left. >> >> I can think of about seven ways to accomplish this, but before I >> dive into >> the code I thought I'd better ask what everyone else has been doing :) >> >> (The only standard way I can think of doing this is to have a SIP >> proxy >> sitting in front of FS the whole time, just to handle these upgrade >> windows. It seems like a bit of a waste.) >> >> So how are you handling your FS software upgrades? >> >> {P^/ >> John >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From johnd at defyne.org Thu Jun 11 11:24:10 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 04:24:10 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> Message-ID: On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote: > On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote: >> >> Exactly. You probably want to have something like this anyways, so that >> when someone accidentally unplugs the system, or the disks/CPU/RAM crash, >> you?re not stuck. >> >> >> >> That is, until FreeSWITCH can record its internal state to some >> inter-machine memory so we can have hot failover. ;) >> >> >> > I think that's going to be in 1.0.5. :) I'm still too much of a noob to be certain that's a joke :) ... but FS core already does record much of its internal state... to a DB, right? It just has to not clear that out on startup and problem solved! OTOH there will be a bit of trouble getting the internal state out of all those modules and libraries... in particular sofia :D {P^/ From larclap at yahoo.com Thu Jun 11 11:27:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 11:27:59 -0700 Subject: [Freeswitch-users] Can't hear outbound calls In-Reply-To: <87f2f3b90906101718g72911b8fv326ee9c470fdf097@mail.gmail.com> References: <005001c9e8a9$76c5c780$64515680$@com> <015a01c9e923$bf458020$3dd08060$@com> <020001c9e948$e2cca7b0$a865f710$@com> <024701c9e95c$07e71ea0$17b55be0$@com> <027501c9e963$69e8fa40$3dbaeec0$@com> <015b01c9ea26$d0abb2e0$720318a0$@com> <87f2f3b90906101718g72911b8fv326ee9c470fdf097@mail.gmail.com> Message-ID: <011001c9eac2$59884e70$0c98eb50$@com> Michael, Removing everything between the tag in sip_profiles/internal/example.xml did the trick - no error message on FS startup. I'm running 13723. 2009-06-11 07:21:03.609317 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-11 07:21:03.612274 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-11 07:21:03.995025 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-11 07:21:03.995436 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-11 07:21:04.245056 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-11 07:21:04.246056 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-11 07:21:04.745950 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-11 07:21:05.745725 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-11 07:21:07.745256 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-11 07:21:12.221251 [DEBUG] switch_nat.c:77 No InternetGatewayDevice, using first entry as default. 2009-06-11 07:21:12.234867 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 10, 2009 5:18 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb wrote: Rupa, I think the console log has information in it that log/freeswitch.log does not. Console: [root at fs bin]# ../freeswitch 2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing Engine. 2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch thread 0 2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml (Invalid or incomplete multibyte or wide character) 2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT 2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5 2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5 2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5 2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5 2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5 2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP 2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT detected! log/freeswitch.log: 2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock interface 'console' to wait for existing references. (from previous Freeswitch invocation) 2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding Dialplan 'enum' 2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding Application 'enum' 2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum' 2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API Function 'enum_auto' 2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default template. 2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql. 2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2. 2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template example. 2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom. 2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template linksys. 2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template asterisk. I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console log. However, the disk log begins at 16:12:58, whereas the console log starts at 16:12:50. The console log finishes its NAT and UPnP reporting before the disk log begins, so I wouldn't see any 0.0.0.0 if it were present. The [ERR] was due to me removing example.xml from sip_profiles/internal. I put it back after this. I don't understand the following command in conf/sofia.conf.xml. I think this is just a cosmetic error. You could probably put an empty xml file in sip_profiles/internal and be done with it. Or possibly have just an empty include node, like "" Try it out and report back - we're dying to know what happens! ;) -MC Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 6:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls if you haven't changed your logging, then it is probably ok. The 0.0.0.0 thing is logged at error level, so will show up in the logs. How did you search? Grep? grep '0\.0\.0\.0' freeswitch.log On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb wrote: Rupa, What options do I have for setting up logging? I'm sorry, but I don't know anything about this. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 09, 2009 5:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't hear outbound calls On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: It looks like the error message only appears on the console when started without the nc option; and it does not appear in log/freeswitch.log in any case. You might want to review how you have your logging setup then. The example I gave you was copied/pasted out of my freeswitch.log file while testing this fix. -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/b45b3d0c/attachment-0002.html From msc at freeswitch.org Thu Jun 11 11:29:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 11:29:15 -0700 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> Message-ID: <87f2f3b90906111129w5f5c5678lf05b025f57447e06@mail.gmail.com> On Thu, Jun 11, 2009 at 11:24 AM, John Dalgliesh wrote: > On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote: > >> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote: >> >>> >>> Exactly. You probably want to have something like this anyways, so that >>> when someone accidentally unplugs the system, or the disks/CPU/RAM crash, >>> you?re not stuck. >>> >>> >>> >>> That is, until FreeSWITCH can record its internal state to some >>> inter-machine memory so we can have hot failover. ;) >>> >>> >>> >>> I think that's going to be in 1.0.5. :) >> > > I'm still too much of a noob to be certain that's a joke :) ... but FS core > already does record much of its internal state... to a DB, right? It just > has to not clear that out on startup and problem solved! > It was a joke. :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/e9bf3fbf/attachment-0002.html From mike at jerris.com Thu Jun 11 11:30:57 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Jun 2009 14:30:57 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C670262E23E56@mse17be1.mse17.exchange.ms> <87f2f3b90906111042j3f8b5992tc05b65bb9256245c@mail.gmail.com> Message-ID: <74170625-2AA0-4FF6-969D-3C850DCD7CA0@jerris.com> On Jun 11, 2009, at 2:24 PM, John Dalgliesh wrote: > On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote: >> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote: >>> >>> Exactly. You probably want to have something like this anyways, so >>> that >>> when someone accidentally unplugs the system, or the disks/CPU/RAM >>> crash, >>> you?re not stuck. >>> >>> That is, until FreeSWITCH can record its internal state to some >>> inter-machine memory so we can have hot failover. ;) >>> >> I think that's going to be in 1.0.5. :) > > I'm still too much of a noob to be certain that's a joke :) ... but > FS core already does record much of its internal state... to a DB, > right? It just has to not clear that out on startup and problem > solved! > > OTOH there will be a bit of trouble getting the internal state out > of all those modules and libraries... in particular sofia :D We have talked quite some about this, its a major job, easily months of work for multiple programmers. We would love to do it but its not on any roadmaps at this time. Mike From brian at freeswitch.org Thu Jun 11 11:07:14 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 13:07:14 -0500 Subject: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition In-Reply-To: <00d501c9eabf$03c7ab00$0b570100$@com> References: <00d501c9eabf$03c7ab00$0b570100$@com> Message-ID: <6603D022-5AD5-4416-9978-8F0E20AAD3A8@freeswitch.org> try destination_number /b On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote: > http://pastebin.freeswitch.org/9365 > > I do not know what I am doing wrong. I am trying to set the > effective_caller_id_name and _number depending on the originating > extension. > > I tried: > expression="^100[09]$" break="on-true"> > and > expression="^100[09]$" break="on-true"> > and > expression="^100[09]$" break="on-true"> > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/7c568749/attachment-0002.html From mrene_lists at avgs.ca Thu Jun 11 11:38:10 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 11 Jun 2009 14:38:10 -0400 Subject: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition In-Reply-To: <6603D022-5AD5-4416-9978-8F0E20AAD3A8@freeswitch.org> References: <00d501c9eabf$03c7ab00$0b570100$@com> <6603D022-5AD5-4416-9978-8F0E20AAD3A8@freeswitch.org> Message-ID: Your syntax is also wrong, it should be and NOT field=${varname}$ Math On 11-Jun-09, at 2:07 PM, Brian West wrote: > try destination_number > > /b > > On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote: > >> http://pastebin.freeswitch.org/9365 >> >> I do not know what I am doing wrong. I am trying to set the >> effective_caller_id_name and _number depending on the originating >> extension. >> >> I tried: >> > expression="^100[09]$" break="on-true"> >> and >> > expression="^100[09]$" break="on-true"> >> and >> > expression="^100[09]$" break="on-true"> >> > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/8ca17349/attachment-0002.html From kristian.kielhofner at gmail.com Thu Jun 11 11:41:51 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 11 Jun 2009 14:41:51 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> References: <36574815-1F9B-4724-B587-C258687D2BD0@halo2.pl> <191c3a030906110954k4e71d59es5d16544cf1669f0b@mail.gmail.com> Message-ID: <2d9149cd0906111141m66a7fe1ao6bc29f520fb08f06@mail.gmail.com> That's exactly what I do. Between dispatcher and FLAGS/GFLAGS this is easy to do in OpenSIPS/SER. On Thu, Jun 11, 2009 at 12:54 PM, Anthony Minessale wrote: > or you can put a sip proxy in front of 2 boxes where you can control the > flow of traffic. > when you want to upgrade one, take all the traffic off of it by forcing all > calls to the other box, upgrade it then shift the traffic to the new one. > if that goes well, upgrade the other one too. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From larclap at yahoo.com Thu Jun 11 12:49:24 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 12:49:24 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match Message-ID: <016601c9eacd$b8ee4cb0$2acae610$@com> I have a match expression for outbound calls as "\d{10}". It's fine for unformatted numbers. Not knowing any better, I created another extension to handle numbers formatted like XXX-XXX-XXXX, which is easier to read and exists in one hard phone's phonebook. It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many extensions for different formats. There's got to be a better way. Any suggestions? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/ebb49988/attachment-0002.html From brian at freeswitch.org Thu Jun 11 12:57:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 14:57:52 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <016601c9eacd$b8ee4cb0$2acae610$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> Message-ID: <3ED39806-9163-4593-B1AE-E4655EAF625B@freeswitch.org> Don't do it! Doing that stuff is highly silly. /b On Jun 11, 2009, at 2:49 PM, Lars Zeb wrote: > There?s got to be a better way. Any suggestions? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/a57d347b/attachment-0002.html From enno.egbert at web.de Thu Jun 11 13:04:40 2009 From: enno.egbert at web.de (NOx-WHV) Date: Thu, 11 Jun 2009 13:04:40 -0700 (PDT) Subject: [Freeswitch-users] re direct calls Message-ID: <23982889.post@talk.nabble.com> Hello Freeswitch User! I am using FS since a few weeks. My intent is to have clients who uses TLS and SRTP for a full encrypted call. I just managed it, that calls are encrypted with TLS and SRTP. My second aim ist to redirect this calls for reduce the processing of the server. I only use the FS for calls between users of this freeswitch. I just tested a "redirect" in the dialplan (without TLS and SRTP) but it doesn?t work. How i have to configure the dialplan for redirect the call to the other user. I use a SNOM hardphone and a phonerlite softphone. Thanks for sour help! NOX -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23982889.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Jun 11 13:11:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 15:11:09 -0500 Subject: [Freeswitch-users] re direct calls In-Reply-To: <23982889.post@talk.nabble.com> References: <23982889.post@talk.nabble.com> Message-ID: On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote: > > Hello Freeswitch User! > > I am using FS since a few weeks. My intent is to have clients who > uses TLS > and SRTP for a full encrypted call. > > I just managed it, that calls are encrypted with TLS and SRTP. My > second aim > ist to redirect this calls for reduce the processing of the server. > I only > use the FS for calls between users of this freeswitch. I just tested a > "redirect" in the dialplan (without TLS and SRTP) but it doesn?t > work. How i > have to configure the dialplan for redirect the call to the other > user. You can't. Its not possible because we are a b2bua and you have already negotiated the keys between the endpoints and FreeSWITCH and when you redirect the media neither phone can decrypt the packets correctly. > > I use a SNOM hardphone and a phonerlite softphone. > > Thanks for sour help! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/bcc62b66/attachment-0002.html From mgg at giagnocavo.net Thu Jun 11 13:33:04 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 11 Jun 2009 16:33:04 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Well, if you're running multiple machines, waiting for it to drainstop isn't that big of a deal unless you're in some sort of hurry, right? Give it an hour or so to drainstop, then kill 'em. Would it not be simpler to try to do something with re-invites or REFER, assuming your endpoints support it? -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Dalgliesh Sent: Thursday, June 11, 2009 12:14 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques I assume he's talking about hardware failures here :P But to answer the question: crashes are easy to deal with. With a crash you have lost the calls that are in progress anyway; you don't have to manage a gradual transition. Currently, since FS is quite quick to start up, I am just relaunching it immediately. But when I have a second box up and running what I'll do is just add the IP of the dead machine as another IP of the second box, and then it will take all the old machine's traffic. That is the plan anyway. I've seen some commercial boxes that use a similar trick. (I've only seen one crash that wasn't my fault. Something to do with terminating a bridge: when the first leg gets a hangup it hangs up the other leg on its own thread... which can cause problems if the other leg was doing something funky at the time. Leads to a heap corruption. Doesn't happen with MALLOC_CHECK_ set so I'm just leaving it set for now :) {P^/ On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote: > > By reporting it on Jira so it doesn't crash anymore :D > > > On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote: > >> How are you handling your FS box crashing? >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of John Dalgliesh >> Sent: Wednesday, June 10, 2009 9:04 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Live Upgrade Techniques >> >> >> Hi, >> >> I am slowly gaining confidence using FreeSWITCH in production, but >> there >> is one issue that I'm still wondering about: how are people upgrading >> their FreeSWITCH installation binaries without dropping all current >> calls? >> >> So far I have been upgrading in the dead of night, after pausing for 5 >> minutes then dropping the stragglers, but this is hardly ideal. >> >> What I would like to do is to run an upgraded instance of FreeSWITCH >> on >> the same machine, and have it handle all new call packets, whereas >> the old >> instance continues to handle the existing call packets, until there >> are no >> more old calls left. >> >> I can think of about seven ways to accomplish this, but before I >> dive into >> the code I thought I'd better ask what everyone else has been doing :) >> >> (The only standard way I can think of doing this is to have a SIP >> proxy >> sitting in front of FS the whole time, just to handle these upgrade >> windows. It seems like a bit of a waste.) >> >> So how are you handling your FS software upgrades? >> >> {P^/ >> John >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:07:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:07:59 +0100 Subject: [Freeswitch-users] Orphaned calls Message-ID: Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,, ,,,,, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/59562fa1/attachment-0002.html From enno.egbert at web.de Thu Jun 11 14:16:07 2009 From: enno.egbert at web.de (NOx-WHV) Date: Thu, 11 Jun 2009 14:16:07 -0700 (PDT) Subject: [Freeswitch-users] re direct calls In-Reply-To: References: <23982889.post@talk.nabble.com> Message-ID: <23989162.post@talk.nabble.com> Thanks for your answer. Can you just announce b2bua. Brian West-3 wrote: > > > On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote: > >> >> Hello Freeswitch User! >> >> I am using FS since a few weeks. My intent is to have clients who >> uses TLS >> and SRTP for a full encrypted call. >> >> I just managed it, that calls are encrypted with TLS and SRTP. My >> second aim >> ist to redirect this calls for reduce the processing of the server. >> I only >> use the FS for calls between users of this freeswitch. I just tested a >> "redirect" in the dialplan (without TLS and SRTP) but it doesn?t >> work. How i >> have to configure the dialplan for redirect the call to the other >> user. > > You can't. Its not possible because we are a b2bua and you have > already negotiated the keys between the endpoints and FreeSWITCH and > when you redirect the media neither phone can decrypt the packets > correctly. > >> >> I use a SNOM hardphone and a phonerlite softphone. >> >> Thanks for sour help! > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23989162.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From john at feith.com Thu Jun 11 14:17:34 2009 From: john at feith.com (John Wehle) Date: Thu, 11 Jun 2009 17:17:34 -0400 (EDT) Subject: [Freeswitch-users] Caller id when doing transfers Message-ID: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> > It appears from some limited testing that the original caller id is always > shown when the call is transfered. Is there some way to have the person > making the transfer show up as the caller id? To answer my own question it appears that the information is available in the sip_h_Referred-By variable. E.g.: allows the station id making the transfer to be known when a call is transfered to *5. The station id can then be used to park the call in the proper fifo. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From msc at freeswitch.org Thu Jun 11 14:20:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 14:20:13 -0700 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: References: Message-ID: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > > > Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 > sessions remaining like the one below (number and ISP changed) > > > > Anyone have an idea why these 6 sessions remain? I also had 120 calls > that I didn?t get a hang-up for, but that might be me not processing the > events fast enough. > > > Do they show on "show calls"? Or do they show up on "show channels" only? Just curious to see if they were bridged or not. -MC > That said, FS was handling a steady concurrent call level of around 350 > which was awesome !! > > > > Regards > > > > > > UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 > microseconds > > 86913 session(s) since startup > > > > f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 > 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net > ,CS_NEW,,,,,,,,,,,,, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/a27bc751/attachment-0002.html From msc at freeswitch.org Thu Jun 11 14:20:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 14:20:52 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <016601c9eacd$b8ee4cb0$2acae610$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> Message-ID: <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb wrote: > I have a match expression for outbound calls as ?\d{10}?. It?s fine for > unformatted numbers. Not knowing any better, I created another extension to > handle numbers formatted like XXX-XXX-XXXX, which is easier to read and > exists in one hard phone?s phonebook. > > > > It looks like: ?^1?(\d{3})-(\d{3})-(\d{4})$?. But I can see making many > extensions for different formats. > Out of curiosity, what benefit does having all these formats get you? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/1a0f1b8d/attachment-0002.html From msc at freeswitch.org Thu Jun 11 14:21:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 11 Jun 2009 14:21:57 -0700 Subject: [Freeswitch-users] Caller id when doing transfers In-Reply-To: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> References: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> Message-ID: <87f2f3b90906111421u5781c7bbob3704486f745a6cc@mail.gmail.com> On Thu, Jun 11, 2009 at 2:17 PM, John Wehle wrote: > > It appears from some limited testing that the original caller id is > always > > shown when the call is transfered. Is there some way to have the person > > making the transfer show up as the caller id? > > To answer my own question it appears that the information is available > in the sip_h_Referred-By variable. E.g.: > > > > expression="^ > > allows the station id making the transfer to be known when a call is > transfered to *5. The station id can then be used to park the call in > the proper fifo. John, Would you be willing to add this wonderful knowledge to the wiki? :) Let me know if you have any questions about where/how to add it and we'll come up with something. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/302d3747/attachment-0002.html From brian at freeswitch.org Thu Jun 11 14:25:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 16:25:12 -0500 Subject: [Freeswitch-users] re direct calls In-Reply-To: <23989162.post@talk.nabble.com> References: <23982889.post@talk.nabble.com> <23989162.post@talk.nabble.com> Message-ID: <38290DC6-93F2-4104-9A60-951ECE54A68F@freeswitch.org> what? On Jun 11, 2009, at 4:16 PM, NOx-WHV wrote: > Can you just announce b2bua. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/93dc9396/attachment-0002.html From brian at freeswitch.org Thu Jun 11 14:26:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 16:26:29 -0500 Subject: [Freeswitch-users] Caller id when doing transfers In-Reply-To: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> References: <200906112117.n5BLHYDd007499@jwlab.FEITH.COM> Message-ID: It does if you do a blind transfer... if you're talking attended transfers thats a whole different ball of wax... /b On Jun 11, 2009, at 4:17 PM, John Wehle wrote: >> It appears from some limited testing that the original caller id is >> always >> shown when the call is transfered. Is there some way to have the >> person >> making the transfer show up as the caller id? > > To answer my own question it appears that the information is available > in the sip_h_Referred-By variable. E.g.: > > > > > > allows the station id making the transfer to be known when a call is > transfered to *5. The station id can then be used to park the call in > the proper fifo. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/aaa89b77/attachment-0002.html From brian at freeswitch.org Thu Jun 11 14:27:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 16:27:58 -0500 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> Message-ID: You could also attach to it with GDB and see if its hanging somewhere else. /b On Jun 11, 2009, at 4:20 PM, Michael Collins wrote: > Do they show on "show calls"? Or do they show up on "show channels" > only? Just curious to see if they were bridged or not. > -MC Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/ed18bc1c/attachment-0002.html From enno.egbert at web.de Thu Jun 11 14:39:40 2009 From: enno.egbert at web.de (NOx-WHV) Date: Thu, 11 Jun 2009 14:39:40 -0700 (PDT) Subject: [Freeswitch-users] re direct calls In-Reply-To: <23989162.post@talk.nabble.com> References: <23982889.post@talk.nabble.com> <23989162.post@talk.nabble.com> Message-ID: <23989451.post@talk.nabble.com> back to back user agent! :-) Thanks! I just ask google! NOx-WHV wrote: > > Thanks for your answer. > > Can you just announce b2bua. > > > > Brian West-3 wrote: >> >> >> On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote: >> >>> >>> Hello Freeswitch User! >>> >>> I am using FS since a few weeks. My intent is to have clients who >>> uses TLS >>> and SRTP for a full encrypted call. >>> >>> I just managed it, that calls are encrypted with TLS and SRTP. My >>> second aim >>> ist to redirect this calls for reduce the processing of the server. >>> I only >>> use the FS for calls between users of this freeswitch. I just tested a >>> "redirect" in the dialplan (without TLS and SRTP) but it doesn?t >>> work. How i >>> have to configure the dialplan for redirect the call to the other >>> user. >> >> You can't. Its not possible because we are a b2bua and you have >> already negotiated the keys between the endpoints and FreeSWITCH and >> when you redirect the media neither phone can decrypt the packets >> correctly. >> >>> >>> I use a SNOM hardphone and a phonerlite softphone. >>> >>> Thanks for sour help! >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/redirect-calls-tp23982889p23989451.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Jun 11 14:40:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 11 Jun 2009 16:40:17 -0500 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: References: Message-ID: <191c3a030906111440g7c60a82dt1e4eab5ba6e45857@mail.gmail.com> it may be a race in the sql event handler where the delete comes before the insert on a really short call. how many sessions did "status" report were in use? On Thu, Jun 11, 2009 at 4:07 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > > > Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 > sessions remaining like the one below (number and ISP changed) > > > > Anyone have an idea why these 6 sessions remain? I also had 120 calls > that I didn?t get a hang-up for, but that might be me not processing the > events fast enough. > > > > That said, FS was handling a steady concurrent call level of around 350 > which was awesome !! > > > > Regards > > > > > > UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 > microseconds > > 86913 session(s) since startup > > > > f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 > 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net > ,CS_NEW,,,,,,,,,,,,, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/7ba1cfca/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:40:43 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:40:43 +0100 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> Message-ID: It was the output from show channels. I've rebooted the server now, so I can't run show calls. I'll see what happens tomorrow. Certainly running status showed 6 sessions All calls are initiated using and 'Originate' from an inbound socket Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 11 June 2009 22:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton wrote: Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. Do they show on "show calls"? Or do they show up on "show channels" only? Just curious to see if they were bridged or not. -MC That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,, ,,,,, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/d9de3a1c/attachment-0002.html From mike at jerris.com Thu Jun 11 14:50:42 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 11 Jun 2009 17:50:42 -0400 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> Message-ID: <1E7EA5F8-FAFD-40FD-B7CC-22B396DC5B3B@jerris.com> If they were still showing in status, can you use gcore to dump a core next time this happens, leave it running somewhere we can get to it and post a thread apply all bt to Jira. Mike On Jun 11, 2009, at 5:40 PM, "Nik Middleton" wrote: > It was the output from show channels. I?ve rebooted the server now, > so I can?t run show calls. I?ll see what happens tomorrow. Certai > nly running status showed 6 sessions > > > > All calls are initiated using and ?Originate? from an inbound socket > > > > Regards > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: 11 June 2009 22:20 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Orphaned calls > > > > > > On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton > wrote: > > > > Ok, so I did a mere 86,000 calls today, but when it was all over, I > had 6 sessions remaining like the one below (number and ISP changed) > > > > Anyone have an idea why these 6 sessions remain? I also had 120 > calls that I didn?t get a hang-up for, but that might be me not proc > essing the events fast enough. > > > > Do they show on "show calls"? Or do they show up on "show channels" > only? Just curious to see if they were bridged or not. > -MC > > That said, FS was handling a steady concurrent call level of around > 350 which was awesome !! > > > > Regards > > > > > > UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 > milliseconds, 400 microseconds > > 86913 session(s) since startup > > > > f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 > 18:38:23,1244741903,sofia/external/ > 0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,,,,,,, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/3b1b1ea1/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:57:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:57:33 +0100 Subject: [Freeswitch-users] Status Event Message-ID: Not sure where enhancement requests should be posted, but here it is anyway I would dearly love to be able to send a status event that returns an event style output that provides machine readable output rather than the wordy human readable response. (I hate parsing) Is there such an event already? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/30ba894d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Jun 11 14:58:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 11 Jun 2009 22:58:51 +0100 Subject: [Freeswitch-users] Orphaned calls In-Reply-To: <1E7EA5F8-FAFD-40FD-B7CC-22B396DC5B3B@jerris.com> References: <87f2f3b90906111420p54744199h2b7e1c2a97763f81@mail.gmail.com> <1E7EA5F8-FAFD-40FD-B7CC-22B396DC5B3B@jerris.com> Message-ID: Will do Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 11 June 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls If they were still showing in status, can you use gcore to dump a core next time this happens, leave it running somewhere we can get to it and post a thread apply all bt to Jira. Mike On Jun 11, 2009, at 5:40 PM, "Nik Middleton" wrote: It was the output from show channels. I've rebooted the server now, so I can't run show calls. I'll see what happens tomorrow. Certainly running status showed 6 sessions All calls are initiated using and 'Originate' from an inbound socket Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 11 June 2009 22:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6 sessions remaining like the one below (number and ISP changed) Anyone have an idea why these 6 sessions remain? I also had 120 calls that I didn't get a hang-up for, but that might be me not processing the events fast enough. Do they show on "show calls"? Or do they show up on "show channels" only? Just curious to see if they were bridged or not. -MC That said, FS was handling a steady concurrent call level of around 350 which was awesome !! Regards UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400 microseconds 86913 session(s) since startup f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11 18:38:23,1244741903,sofia/external/ 0XXXXXXXX at gk.myISP.net,CS_NEW,,,,,,,,,,,,, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c51d0134/attachment-0002.html From larclap at yahoo.com Thu Jun 11 15:57:52 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 15:57:52 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> Message-ID: <01fc01c9eae8$0cf317e0$26d947a0$@com> The users entering numbers into their phonebooks are able to recognize the number more easily. I will tell them to forget it and make the phone numbers numeric only. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, June 11, 2009 2:21 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb wrote: I have a match expression for outbound calls as "\d{10}". It's fine for unformatted numbers. Not knowing any better, I created another extension to handle numbers formatted like XXX-XXX-XXXX, which is easier to read and exists in one hard phone's phonebook. It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many extensions for different formats. Out of curiosity, what benefit does having all these formats get you? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/556cec3e/attachment-0002.html From brian at freeswitch.org Thu Jun 11 16:04:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 18:04:57 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <01fc01c9eae8$0cf317e0$26d947a0$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> Message-ID: <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> If the phone sends them with dashes in them the phone IS BROKEN and should be smashed with a hammer. /b On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote: > The users entering numbers into their phonebooks are able to > recognize the number more easily. > > I will tell them to forget it and make the phone numbers numeric only. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/e32cc907/attachment-0002.html From jmesquita at gmail.com Thu Jun 11 16:25:05 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 11 Jun 2009 20:25:05 -0300 Subject: [Freeswitch-users] Status Event In-Reply-To: References: Message-ID: <5a8712120906111625o46598794ica826e07001d4dcc@mail.gmail.com> Nik, I am a noobie and all, but most API responses can come as xml just by adding "as xml" at the end of the call. jmesquita On Thu, Jun 11, 2009 at 6:57 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Not sure where enhancement requests should be posted, but here it is > anyway > > > > > > I would dearly love to be able to send a status event that returns an event > style output that provides machine readable output rather than the wordy > human readable response. (I hate parsing) > > > > Is there such an event already? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/02b5295f/attachment-0002.html From brian at freeswitch.org Thu Jun 11 16:35:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 18:35:00 -0500 Subject: [Freeswitch-users] Status Event In-Reply-To: <5a8712120906111625o46598794ica826e07001d4dcc@mail.gmail.com> References: <5a8712120906111625o46598794ica826e07001d4dcc@mail.gmail.com> Message-ID: <8CE37A09-F0B2-47F5-94AE-CDB74483A1FF@freeswitch.org> Only if they have an as xml modifier /b On Jun 11, 2009, at 6:25 PM, Jo?o Mesquita wrote: > Nik, I am a noobie and all, but most API responses can come as xml > just by adding "as xml" at the end of the call. > > jmesquita Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/4a00a732/attachment-0002.html From larclap at yahoo.com Thu Jun 11 17:30:39 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 17:30:39 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> Message-ID: <021f01c9eaf5$033e2020$09ba6060$@com> It's a SNOM 320. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 4:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match If the phone sends them with dashes in them the phone IS BROKEN and should be smashed with a hammer. /b On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote: The users entering numbers into their phonebooks are able to recognize the number more easily. I will tell them to forget it and make the phone numbers numeric only. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/b1282488/attachment-0002.html From brian at freeswitch.org Thu Jun 11 17:39:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 19:39:43 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <021f01c9eaf5$033e2020$09ba6060$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> Message-ID: <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> What firmware? /b On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote: > It?s a SNOM 320. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c0938532/attachment-0002.html From larclap at yahoo.com Thu Jun 11 18:34:14 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 18:34:14 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> Message-ID: <023301c9eafd$e509f3f0$af1ddbd0$@com> snom320-SIP 6.5.17. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 5:40 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match What firmware? /b On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote: It's a SNOM 320. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/ec2de897/attachment-0002.html From brian at freeswitch.org Thu Jun 11 18:41:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 20:41:12 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <023301c9eafd$e509f3f0$af1ddbd0$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> <023301c9eafd$e509f3f0$af1ddbd0$@com> Message-ID: <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> You should be running 7.1.35 or higher. /b On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote: > snom320-SIP 6.5.17. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/aef7261a/attachment-0002.html From larclap at yahoo.com Thu Jun 11 19:21:46 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 11 Jun 2009 19:21:46 -0700 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> <023301c9eafd$e509f3f0$af1ddbd0$@com> <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> Message-ID: <024401c9eb04$89538b00$9bfaa100$@com> snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX-XXXX but delivers XXXXXXXXXX to FS. Thanks Brian From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, June 11, 2009 6:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan XML phone number match You should be running 7.1.35 or higher. /b On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote: snom320-SIP 6.5.17. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/1a6968d6/attachment-0002.html From brian at freeswitch.org Thu Jun 11 19:27:23 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 21:27:23 -0500 Subject: [Freeswitch-users] Dialplan XML phone number match In-Reply-To: <024401c9eb04$89538b00$9bfaa100$@com> References: <016601c9eacd$b8ee4cb0$2acae610$@com> <87f2f3b90906111420u21c67638nb576d9e56776c47a@mail.gmail.com> <01fc01c9eae8$0cf317e0$26d947a0$@com> <6FE06C9B-83DE-42DB-80BD-1DC8AF72DADE@freeswitch.org> <021f01c9eaf5$033e2020$09ba6060$@com> <6426BC36-C56D-4CC9-8531-2B3630EA0CFA@freeswitch.org> <023301c9eafd$e509f3f0$af1ddbd0$@com> <413985EA-87BF-4519-BD75-81955908F812@freeswitch.org> <024401c9eb04$89538b00$9bfaa100$@com> Message-ID: See I knew that was a bit of crack :P, Good to hear its working like it SHOULD now! /b On Jun 11, 2009, at 9:21 PM, Lars Zeb wrote: > snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX-XXXX > but delivers XXXXXXXXXX to FS. > > Thanks Brian Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/c96ea677/attachment-0002.html From johnd at defyne.org Thu Jun 11 20:35:58 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 13:35:58 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Message-ID: Hi, On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > > Well, if you're running multiple machines, waiting for it to drainstop > isn't that big of a deal unless you're in some sort of hurry, right? > Give it an hour or so to drainstop, then kill 'em. Yes that's exactly what I'm trying to do. The problem is some people will only try one IP address. > Would it not be simpler to try to do something with re-invites or REFER, > assuming your endpoints support it? That was actually plan A. I already added a property in sip_profile called failover_redirect, which specifies another server to try if FS can't allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), by sending back a SIP 302 Moved Temporarily response, instead of 503 Max Calls In Progress. Turns out not all my endpoints support it :( I considered REFER too but there seems to be even less support for that. If I can't get the socket-sharing upgrade working then I will fall back to this - and peers which don't support the 302 response (or more likely, don't authorise it) will just get no service during the upgrade. > -Michael {P^/ > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Dalgliesh > Sent: Thursday, June 11, 2009 12:14 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Live Upgrade Techniques > > > I assume he's talking about hardware failures here :P > > But to answer the question: crashes are easy to deal with. With a crash > you have lost the calls that are in progress anyway; you don't have to > manage a gradual transition. > > Currently, since FS is quite quick to start up, I am just relaunching it > immediately. > > But when I have a second box up and running what I'll do is just add the > IP of the dead machine as another IP of the second box, and then it will > take all the old machine's traffic. That is the plan anyway. I've seen > some commercial boxes that use a similar trick. > ... From brian at freeswitch.org Thu Jun 11 20:57:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 11 Jun 2009 22:57:37 -0500 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Message-ID: <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > > Hi, > > On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >> >> Well, if you're running multiple machines, waiting for it to >> drainstop >> isn't that big of a deal unless you're in some sort of hurry, right? >> Give it an hour or so to drainstop, then kill 'em. > > Yes that's exactly what I'm trying to do. The problem is some people > will > only try one IP address. Clients that don't properly implement SRV/NAPTR and fail over need to be smacked. :) (not customers but software that fails to do that) > >> Would it not be simpler to try to do something with re-invites or >> REFER, >> assuming your endpoints support it? > > That was actually plan A. I already added a property in sip_profile > called > failover_redirect, which specifies another server to try if FS can't > allocate any more sessions (e.g. too busy, paused, shutdown asap, > etc.), > by sending back a SIP 302 Moved Temporarily response, instead of 503 > Max > Calls In Progress. You can't send a 302 to a call thats already established. > > Turns out not all my endpoints support it :( AKA broken endpoints. :) > > I considered REFER too but there seems to be even less support for > that. ACK really? thats sad! > > If I can't get the socket-sharing upgrade working then I will fall > back to > this - and peers which don't support the 302 response (or more likely, > don't authorise it) will just get no service during the upgrade. > >> -Michael > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090611/04f3ccda/attachment-0002.html From mgg at giagnocavo.net Thu Jun 11 21:16:51 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 12 Jun 2009 00:16:51 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E23F60@mse17be1.mse17.exchange.ms> >> Well, if you're running multiple machines, waiting for it to drainstop >> isn't that big of a deal unless you're in some sort of hurry, right? >> Give it an hour or so to drainstop, then kill 'em. > >Yes that's exactly what I'm trying to do. The problem is some people will >only try one IP address. Right, so if you have a proxy in front that is handling this, it should be no problem. From shahzad at vopium.com Thu Jun 11 21:45:57 2009 From: shahzad at vopium.com (Muhammad Shahzad) Date: Fri, 12 Jun 2009 10:45:57 +0600 Subject: [Freeswitch-users] MPL Confusion Message-ID: Hi, I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do understand that me or any one can change provided code according to our customization needs and we are not bound to share our changes as long as we are not distributing it, right? Now, i have been doing R&D on MSN and Yahoo voice chat services, I have now completed by research and now would like to write up FS modules to communicate with these servers. But as you all know both MSN and Yahoo provide SIP based VOIP services, however they are not using standard SIP stack and have their own versions of customized SIP stack. So, in order to write an endpoint for these servers, instead of writing everything from scretch, i can using existing mod_sofia endpoint and customize it to make it compatible with MSN and Yahoo SIP stack. So here are my questions, 1. Is it possible under MPL, that i make a copy of mod_sofia as say mod_msn and develop it to work with MSN, similarly mod_yahoo for Yahoo voice chat service? 2. If yes, how can i mention my role in these modules development, i.e. as developer or as contributor? Also i wish to include my work, once completed, in FreeSWITCH, can you provide me the guidelines and / or eligibility criteria to do so, any link on FS site etc.? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/f9d2d154/attachment-0002.html From johnd at defyne.org Thu Jun 11 23:25:58 2009 From: johnd at defyne.org (John Dalgliesh) Date: Fri, 12 Jun 2009 16:25:58 +1000 (EST) Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> Message-ID: On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >>> >>> Well, if you're running multiple machines, waiting for it to drainstop >>> isn't that big of a deal unless you're in some sort of hurry, right? >>> Give it an hour or so to drainstop, then kill 'em. >> >> Yes that's exactly what I'm trying to do. The problem is some people will >> only try one IP address. > > Clients that don't properly implement SRV/NAPTR and fail over need to be > smacked. :) (not customers but software that fails to do that) Yes I'm sure much of their software can do this but it has been set up for static numeric IPs. And getting the IP changed is a week-long process for some customers! >>> Would it not be simpler to try to do something with re-invites or REFER, >>> assuming your endpoints support it? >> >> That was actually plan A. I already added a property in sip_profile called >> failover_redirect, which specifies another server to try if FS can't >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max >> Calls In Progress. > > You can't send a 302 to a call thats already established. Yes and I don't want to touch established calls - those calls can stay there until they drop. This is sent to new requests when switch_core_session_request fails in mod_sofia. >> Turns out not all my endpoints support it :( > > AKA broken endpoints. :) Some are broken. Some just have this feature disabled. For 'security reasons'. You know the drill. {P^/ John From saeedahmad1981 at gmail.com Fri Jun 12 03:16:27 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 12 Jun 2009 12:16:27 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> Message-ID: I've experience with a commercial SBC, these are two machines running in cluster mode. In that case if one SBC is going down then other will take all new calls including the call which were active on broken SBC (SIP only). Thats quite ideal for wholesale traffic where the SBC will never be idle. On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh wrote: > On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > >>> > >>> Well, if you're running multiple machines, waiting for it to drainstop > >>> isn't that big of a deal unless you're in some sort of hurry, right? > >>> Give it an hour or so to drainstop, then kill 'em. > >> > >> Yes that's exactly what I'm trying to do. The problem is some people > will > >> only try one IP address. > > > > Clients that don't properly implement SRV/NAPTR and fail over need to be > > smacked. :) (not customers but software that fails to do that) > > Yes I'm sure much of their software can do this but it has been set up for > static numeric IPs. And getting the IP changed is a week-long process for > some customers! > > >>> Would it not be simpler to try to do something with re-invites or > REFER, > >>> assuming your endpoints support it? > >> > >> That was actually plan A. I already added a property in sip_profile > called > >> failover_redirect, which specifies another server to try if FS can't > >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), > >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max > >> Calls In Progress. > > > > You can't send a 302 to a call thats already established. > > Yes and I don't want to touch established calls - those calls can stay > there until they drop. This is sent to new requests when > switch_core_session_request fails in mod_sofia. > > >> Turns out not all my endpoints support it :( > > > > AKA broken endpoints. :) > > Some are broken. Some just have this feature disabled. For 'security > reasons'. You know the drill. > > > {P^/ > John > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/b7f3161b/attachment-0002.html From helmut.kuper at ewetel.de Fri Jun 12 05:43:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 12 Jun 2009 14:43:02 +0200 Subject: [Freeswitch-users] Database and "Too many open files" Problem In-Reply-To: <191c3a030906101414j1107e49bk3aa8fb778454451c@mail.gmail.com> References: <4A2FEE39.1030709@ewetel.de> <191c3a030906101414j1107e49bk3aa8fb778454451c@mail.gmail.com> Message-ID: <4A324D56.6080304@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Anthony, yes, that's true. Works well now. Thanks a lot! regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKMk1W4tZeNddg3dwRAhzrAJ45OpGkPkpLEPRw17HUpR3CTaxVVwCcD4/0 TJpI0jZez6uOdETu3OtDbc8= =yWhz -----END PGP SIGNATURE----- From grevenx at me.com Fri Jun 12 06:03:33 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Fri, 12 Jun 2009 15:03:33 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms> <15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca> <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> Message-ID: <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Can you comment some more on how this is configured? Would it be something that could be added to the wiki in the SBC setup page? Best regards, Even Andr? Fiskvik On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: > I've experience with a commercial SBC, these are two machines > running in cluster mode. In that case if one SBC is going down then > other will take all new calls including the call which were active > on broken SBC (SIP only). > > Thats quite ideal for wholesale traffic where the SBC will never be > idle. > > On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh > wrote: > On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > >>> > >>> Well, if you're running multiple machines, waiting for it to > drainstop > >>> isn't that big of a deal unless you're in some sort of hurry, > right? > >>> Give it an hour or so to drainstop, then kill 'em. > >> > >> Yes that's exactly what I'm trying to do. The problem is some > people will > >> only try one IP address. > > > > Clients that don't properly implement SRV/NAPTR and fail over need > to be > > smacked. :) (not customers but software that fails to do that) > > Yes I'm sure much of their software can do this but it has been set > up for > static numeric IPs. And getting the IP changed is a week-long > process for > some customers! > > >>> Would it not be simpler to try to do something with re-invites > or REFER, > >>> assuming your endpoints support it? > >> > >> That was actually plan A. I already added a property in > sip_profile called > >> failover_redirect, which specifies another server to try if FS > can't > >> allocate any more sessions (e.g. too busy, paused, shutdown asap, > etc.), > >> by sending back a SIP 302 Moved Temporarily response, instead of > 503 Max > >> Calls In Progress. > > > > You can't send a 302 to a call thats already established. > > Yes and I don't want to touch established calls - those calls can stay > there until they drop. This is sent to new requests when > switch_core_session_request fails in mod_sofia. > > >> Turns out not all my endpoints support it :( > > > > AKA broken endpoints. :) > > Some are broken. Some just have this feature disabled. For 'security > reasons'. You know the drill. > > > {P^/ > John > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/6fec5bb1/attachment-0002.html From larclap at yahoo.com Fri Jun 12 06:33:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 12 Jun 2009 06:33:12 -0700 Subject: [Freeswitch-users] Unregister extension? Message-ID: <003901c9eb62$551d0f60$ff572e20$@com> How can I unregister a softphone's registration? I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I changed the second one to 1000. Now when I do 'sofia status profile internal' all three show up. How do I get rid of the 1001 extension? I shutdown and restarted FS but that didn't do it. I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is blocking the Polycom at that same extension and that is the reason the Polycom is not showing. Thanks, Lars Registrations: ============================================================================ ===================== Call-ID: 3c267015ab6b-bd6gioq5ytor User: 1010 at 192.168.10.29 Contact: "1010" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1010 Auth-Realm: 192.168.10.29 Call-ID: 3c267015afa6-6v0sw4o3qei3 User: 1001 at 192.168.10.29 Contact: "1001" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1001 Auth-Realm: 192.168.10.29 Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. User: 1019 at 192.168.10.29 Contact: "1019" Agent: Bria Professional release 2.4.3 stamp 50906 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) Host: fs IP: 192.168.10.11 Port: 19040 Auth-User: 1019 Auth-Realm: 192.168.10.29 Call-ID: 3c270d667ff5-47fq2p6n1ou1 User: 1000 at 192.168.10.29 Contact: "1000" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1000 Auth-Realm: 192.168.10.29 ============================================================================ ===================== -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/fea90e20/attachment-0002.html From saeedahmad1981 at gmail.com Fri Jun 12 06:39:27 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 12 Jun 2009 15:39:27 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms><15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca><6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Message-ID: No idea at all, It?s a commercial SBC. I wish if we can have same functionality in FS. - Saeed _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Even Andr? Fiskvik Sent: Friday, June 12, 2009 3:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques Can you comment some more on how this is configured? Would it be something that could be added to the wiki in the SBC setup page? Best regards, Even Andr? Fiskvik On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: I've experience with a commercial SBC, these are two machines running in cluster mode. In that case if one SBC is going down then other will take all new calls including the call which were active on broken SBC (SIP only). Thats quite ideal for wholesale traffic where the SBC will never be idle. On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh wrote: On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >>> >>> Well, if you're running multiple machines, waiting for it to drainstop >>> isn't that big of a deal unless you're in some sort of hurry, right? >>> Give it an hour or so to drainstop, then kill 'em. >> >> Yes that's exactly what I'm trying to do. The problem is some people will >> only try one IP address. > > Clients that don't properly implement SRV/NAPTR and fail over need to be > smacked. :) (not customers but software that fails to do that) Yes I'm sure much of their software can do this but it has been set up for static numeric IPs. And getting the IP changed is a week-long process for some customers! >>> Would it not be simpler to try to do something with re-invites or REFER, >>> assuming your endpoints support it? >> >> That was actually plan A. I already added a property in sip_profile called >> failover_redirect, which specifies another server to try if FS can't >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max >> Calls In Progress. > > You can't send a 302 to a call thats already established. Yes and I don't want to touch established calls - those calls can stay there until they drop. This is sent to new requests when switch_core_session_request fails in mod_sofia. >> Turns out not all my endpoints support it :( > > AKA broken endpoints. :) Some are broken. Some just have this feature disabled. For 'security reasons'. You know the drill. {P^/ John _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/ffb2220c/attachment-0002.html From jmesquita at gmail.com Fri Jun 12 06:47:12 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 12 Jun 2009 10:47:12 -0300 Subject: [Freeswitch-users] Unregister extension? In-Reply-To: <003901c9eb62$551d0f60$ff572e20$@com> References: <003901c9eb62$551d0f60$ff572e20$@com> Message-ID: <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> Lars, don't get me wrong but you have been asking questions that are all answered on the wiki: http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoints Might be a good idea to value the work of lots of ppl who have been documenting by actually using the documentation, no? Sorry if that sounds a bit harsh. jmesquita On Fri, Jun 12, 2009 at 10:33 AM, Lars Zeb wrote: > How can I unregister a softphone?s registration? > > > > I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I > changed the second one to 1000. Now when I do ?sofia status profile > internal? all three show up. How do I get rid of the 1001 extension? I > shutdown and restarted FS but that didn?t do it. > > > > I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is > blocking the Polycom at that same extension and that is the reason the > Polycom is not showing. > > > > Thanks, Lars > > > > > > Registrations: > > > ================================================================================================= > > Call-ID: 3c267015ab6b-bd6gioq5ytor > > User: 1010 at 192.168.10.29 > > Contact: "1010" > > Agent: snom320/7.3.14 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) > > Host: fs > > IP: 192.168.10.104 > > Port: 2048 > > Auth-User: 1010 > > Auth-Realm: 192.168.10.29 > > > > Call-ID: 3c267015afa6-6v0sw4o3qei3 > > User: 1001 at 192.168.10.29 > > Contact: "1001" > > Agent: snom320/7.3.14 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) > > Host: fs > > IP: 192.168.10.104 > > Port: 2048 > > Auth-User: 1001 > > Auth-Realm: 192.168.10.29 > > > > Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. > > User: 1019 at 192.168.10.29 > > Contact: "1019" ;rinstance=5394acb4dfa00c0a> > > Agent: Bria Professional release 2.4.3 stamp 50906 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) > > Host: fs > > IP: 192.168.10.11 > > Port: 19040 > > Auth-User: 1019 > > Auth-Realm: 192.168.10.29 > > > > Call-ID: 3c270d667ff5-47fq2p6n1ou1 > > User: 1000 at 192.168.10.29 > > Contact: "1000" > > Agent: snom320/7.3.14 > > Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) > > Host: fs > > IP: 192.168.10.104 > > Port: 2048 > > Auth-User: 1000 > > Auth-Realm: 192.168.10.29 > > > > > ================================================================================================= > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/58ea5b09/attachment-0002.html From larclap at yahoo.com Fri Jun 12 07:07:52 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 12 Jun 2009 07:07:52 -0700 Subject: [Freeswitch-users] Unregister extension? In-Reply-To: <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> References: <003901c9eb62$551d0f60$ff572e20$@com> <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> Message-ID: <005801c9eb67$2d3c1180$87b43480$@com> No, it?s not too harsh, Jo?o, but I hope not all of my questions were answered on the wiki. I do try to go to the wiki first. I think that my total ignorance of the environment makes it difficult for me to do a search on the wiki or Google. I did try before asking this list. My query to Google was ?Freeswitch unregister?. That was the best I could do given my limited knowledge. Thank you for the help. I?ll learn eventually. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Friday, June 12, 2009 6:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Unregister extension? Lars, don't get me wrong but you have been asking questions that are all answered on the wiki: http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoi nts Might be a good idea to value the work of lots of ppl who have been documenting by actually using the documentation, no? Sorry if that sounds a bit harsh. jmesquita On Fri, Jun 12, 2009 at 10:33 AM, Lars Zeb wrote: How can I unregister a softphone?s registration? I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I changed the second one to 1000. Now when I do ?sofia status profile internal? all three show up. How do I get rid of the 1001 extension? I shutdown and restarted FS but that didn?t do it. I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is blocking the Polycom at that same extension and that is the reason the Polycom is not showing. Thanks, Lars Registrations: ============================================================================ ===================== Call-ID: 3c267015ab6b-bd6gioq5ytor User: 1010 at 192.168.10.29 Contact: "1010" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1010 Auth-Realm: 192.168.10.29 Call-ID: 3c267015afa6-6v0sw4o3qei3 User: 1001 at 192.168.10.29 Contact: "1001" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1001 Auth-Realm: 192.168.10.29 Call-ID: OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. User: 1019 at 192.168.10.29 Contact: "1019" Agent: Bria Professional release 2.4.3 stamp 50906 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) Host: fs IP: 192.168.10.11 Port: 19040 Auth-User: 1019 Auth-Realm: 192.168.10.29 Call-ID: 3c270d667ff5-47fq2p6n1ou1 User: 1000 at 192.168.10.29 Contact: "1000" Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1000 Auth-Realm: 192.168.10.29 ============================================================================ ===================== _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/a67f8ac4/attachment-0002.html From helmut.kuper at ewetel.de Fri Jun 12 07:32:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 12 Jun 2009 16:32:05 +0200 Subject: [Freeswitch-users] Q931 TE State Timer Message-ID: <4A3266E5.2000702@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I just want to let you know, that I started some work on Q.931 TE state timers in current openzap ozmod_isdn stack. The current stack has some problems with freeing ressources when far end doesn't follow Q931 state machine cleanly. On my side I mainly miss a RELEASE after sending DISCONNECT and so m more and more channels are wasted over time. Openzap Q931 stack has some preworked timer and state handling (in form of Q931_UX states). Unfortunately it is disabled in source. I enabled it, added some Q931_UX states to Q931StateTE.c, added a timeout handler to ozmod_isdn.c and tested it with my special problem. In lab missed RELEASEs are now detected and the corresponding channel is freed cleanly - - hurray :) I would like to put my work in openzap trunk as soon as it works stable in production and FS board allows me to do so. I can't promise that I will implement all timers for Q931 TE nor I plan to work on Q931 NT timers. As I said: Just for your information. Have a nice weekend! Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKMmbl4tZeNddg3dwRAvWbAJ43Vex8J3LiVMu9mvs4US23eif19gCeO37l 4CGbk8CA+n+/dMRo5y5A+i0= =ZtvM -----END PGP SIGNATURE----- From andy at fabulous4.co.uk Fri Jun 12 07:37:39 2009 From: andy at fabulous4.co.uk (Andy) Date: Fri, 12 Jun 2009 15:37:39 +0100 Subject: [Freeswitch-users] Sample rate and recordFile Message-ID: <137853E7923C47B7890E39796657719E@D810> Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/a3f1625c/attachment-0002.html From brian at freeswitch.org Fri Jun 12 07:44:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 09:44:54 -0500 Subject: [Freeswitch-users] MPL Confusion In-Reply-To: References: Message-ID: <8DBC35F3-0ADE-4391-BEEB-99609F02A739@freeswitch.org> On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: > Hi, > > I have some confusion about FreeSWITCH's Mozilla Public License 1.1. > I do understand that me or any one can change provided code > according to our customization needs and we are not bound to share > our changes as long as we are not distributing it, right? Not 100% true. > > Now, i have been doing R&D on MSN and Yahoo voice chat services, I > have now completed by research and now would like to write up FS > modules to communicate with these servers. But as you all know both > MSN and Yahoo provide SIP based VOIP services, however they are not > using standard SIP stack and have their own versions of customized > SIP stack. So, in order to write an endpoint for these servers, > instead of writing everything from scretch, i can using existing > mod_sofia endpoint and customize it to make it compatible with MSN > and Yahoo SIP stack. So here are my questions, > > 1. Is it possible under MPL, that i make a copy of mod_sofia as say > mod_msn and develop it to work with MSN, similarly mod_yahoo for > Yahoo voice chat service? Chances are they can be integrated as optional behaviors into mod_sofia. Its best to join #freeswitch and talk to us to see if maybe we can provide you guidance on this process. > 2. If yes, how can i mention my role in these modules development, > i.e. as developer or as contributor? Adding your name to the top of the files is usually the best way. > > Also i wish to include my work, once completed, in FreeSWITCH, can > you provide me the guidelines and / or eligibility criteria to do > so, any link on FS site etc.? > You post your work to our issue tracker http://jira.freeswitch.org > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/bd71a6df/attachment-0002.html From brian at freeswitch.org Fri Jun 12 07:49:12 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 09:49:12 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <137853E7923C47B7890E39796657719E@D810> References: <137853E7923C47B7890E39796657719E@D810> Message-ID: <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> No its not possible yet. Voicemail has an option to record in 11025 but not arbitrary rates for recording otherwise. /b On Jun 12, 2009, at 9:37 AM, Andy wrote: > Hi, > > Sorry but I just can't find this in the documentation. I'm using > recordFile to record incoming messages. I'd like the audio files > produced to be 11025Hz rather than 8kHz is this possible? What > setting do I need to change? > > Many thanks > Andy > ________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/de8767b8/attachment-0002.html From brian at freeswitch.org Fri Jun 12 07:51:02 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 09:51:02 -0500 Subject: [Freeswitch-users] Unregister extension? In-Reply-To: <005801c9eb67$2d3c1180$87b43480$@com> References: <003901c9eb62$551d0f60$ff572e20$@com> <5a8712120906120647i5a9bd338l63864195eb82a268@mail.gmail.com> <005801c9eb67$2d3c1180$87b43480$@com> Message-ID: Its one of the 3000 settings you can change on the identity ... but in your case I am not sure it would have removed the old registration before the new one was registered... but check on the preferences or the identity there is a setting to unregister on reboot. /b PS: sofia profile xxx flush_inbound_reg [call-id] On Jun 12, 2009, at 9:07 AM, Lars Zeb wrote: > No, it?s not too harsh, Jo?o, but I hope not all of my questions > were answered on the wiki. > > I do try to go to the wiki first. I think that my total ignorance of > the environment makes it difficult for me to do a search on the wiki > or Google. I did try before asking this list. My query to Google was > ?Freeswitch unregister?. That was the best I could do given my > limited knowledge. > > Thank you for the help. I?ll learn eventually. > > Lars > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/2262e0e3/attachment-0002.html From santhosh.suthakar at gmail.com Fri Jun 12 00:35:07 2009 From: santhosh.suthakar at gmail.com (Santhosh) Date: Fri, 12 Jun 2009 00:35:07 -0700 Subject: [Freeswitch-users] XML-RPC In-Reply-To: <4a31fa03.14b48c0a.4983.fffff107@mx.google.com> References: <4a31fa03.14b48c0a.4983.fffff107@mx.google.com> Message-ID: <4a320541.25578c0a.1cf4.0f2d@mx.google.com> Hi, Is there any where I can find more documentation on the XML-RPC interface of freeswitch. I am trying to initiate a conference and add in users to it from flex. I would appreciate any insights. Thanks San -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/6aace86e/attachment-0002.html From mike at jerris.com Fri Jun 12 08:34:26 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 12 Jun 2009 11:34:26 -0400 Subject: [Freeswitch-users] XML-RPC In-Reply-To: <4a320541.25578c0a.1cf4.0f2d@mx.google.com> References: <4a31fa03.14b48c0a.4983.fffff107@mx.google.com> <4a320541.25578c0a.1cf4.0f2d@mx.google.com> Message-ID: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC http://wiki.freeswitch.org/wiki/Mod_commands http://wiki.freeswitch.org/wiki/Mod_conference Mike On Jun 12, 2009, at 3:35 AM, Santhosh wrote: > Hi, > > Is there any where I can find more documentation on the XML-RPC > interface of freeswitch. I am trying to initiate a conference and > add in users to it from flex. I would appreciate any insights. > > Thanks > San > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/4551e9cd/attachment-0002.html From anthony.minessale at gmail.com Fri Jun 12 08:50:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 12 Jun 2009 10:50:43 -0500 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A3266E5.2000702@ewetel.de> References: <4A3266E5.2000702@ewetel.de> Message-ID: <191c3a030906120850q23b816e0qe8be6f7ae9ca8cc5@mail.gmail.com> please make sure you stay tuned into #openzap and coordinate with stkn and the other guys doing work on the stack. That way we can make sure we get the best out of the code. On Fri, Jun 12, 2009 at 9:32 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I just want to let you know, that I started some work on Q.931 TE state > timers in current openzap ozmod_isdn stack. The current stack has some > problems with freeing ressources when far end doesn't follow Q931 state > machine cleanly. On my side I mainly miss a RELEASE after sending > DISCONNECT and so m more and more channels are wasted over time. > > Openzap Q931 stack has some preworked timer and state handling (in form > of Q931_UX states). Unfortunately it is disabled in source. I enabled > it, added some Q931_UX states to Q931StateTE.c, added a timeout handler > to ozmod_isdn.c and tested it with my special problem. In lab missed > RELEASEs are now detected and the corresponding channel is freed cleanly > - - hurray :) > > I would like to put my work in openzap trunk as soon as it works stable > in production and FS board allows me to do so. > > I can't promise that I will implement all timers for Q931 TE nor I plan > to work on Q931 NT timers. > > As I said: Just for your information. > > > Have a nice weekend! > > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKMmbl4tZeNddg3dwRAvWbAJ43Vex8J3LiVMu9mvs4US23eif19gCeO37l > 4CGbk8CA+n+/dMRo5y5A+i0= > =ZtvM > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/c86dca0a/attachment-0002.html From anthony.minessale at gmail.com Fri Jun 12 09:02:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 12 Jun 2009 11:02:57 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> Message-ID: <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> Looked easy enough so i added record_sample_rate variable that should influence it for you if you set it in advance. On Fri, Jun 12, 2009 at 9:49 AM, Brian West wrote: > No its not possible yet. Voicemail has an option to record in 11025 but > not arbitrary rates for recording otherwise. > /b > > On Jun 12, 2009, at 9:37 AM, Andy wrote: > > Hi, > > Sorry but I just can't find this in the documentation. I'm using recordFile > to record incoming messages. I'd like the audio files produced to be 11025Hz > rather than 8kHz is this possible? What setting do I need to change? > > Many thanks > Andy > ________ > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/503b3f7e/attachment-0002.html From andy at fabulous4.co.uk Fri Jun 12 09:26:08 2009 From: andy at fabulous4.co.uk (Andy) Date: Fri, 12 Jun 2009 17:26:08 +0100 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> Message-ID: Excellent, thanks Anthony, I'll give it a go. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 12 June 2009 17:03 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile Looked easy enough so i added record_sample_rate variable that should influence it for you if you set it in advance. On Fri, Jun 12, 2009 at 9:49 AM, Brian West wrote: No its not possible yet. Voicemail has an option to record in 11025 but not arbitrary rates for recording otherwise. /b On Jun 12, 2009, at 9:37 AM, Andy wrote: Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy ________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/882e9c45/attachment-0002.html From mgg at giagnocavo.net Fri Jun 12 09:54:04 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 12 Jun 2009 12:54:04 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms><15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca><6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> Well, Nextone for instance has a database the keeps most of the state of calls, and it's replicated between the two nodes. (I seem to recall the database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still get truncated when there's any kind of switchover. But Nextone is a closed system with limited services. As MikeJ mentioned, it was discussed for FS, but it's a LOT of work to get that state synchronized. And, every custom app/module would have to register and support recreating their state. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Saeed Ahmed Sent: Friday, June 12, 2009 7:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques No idea at all, It's a commercial SBC. I wish if we can have same functionality in FS. - Saeed ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Even Andr? Fiskvik Sent: Friday, June 12, 2009 3:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques Can you comment some more on how this is configured? Would it be something that could be added to the wiki in the SBC setup page? Best regards, Even Andr? Fiskvik On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: I've experience with a commercial SBC, these are two machines running in cluster mode. In that case if one SBC is going down then other will take all new calls including the call which were active on broken SBC (SIP only). Thats quite ideal for wholesale traffic where the SBC will never be idle. On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh > wrote: On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: >>> >>> Well, if you're running multiple machines, waiting for it to drainstop >>> isn't that big of a deal unless you're in some sort of hurry, right? >>> Give it an hour or so to drainstop, then kill 'em. >> >> Yes that's exactly what I'm trying to do. The problem is some people will >> only try one IP address. > > Clients that don't properly implement SRV/NAPTR and fail over need to be > smacked. :) (not customers but software that fails to do that) Yes I'm sure much of their software can do this but it has been set up for static numeric IPs. And getting the IP changed is a week-long process for some customers! >>> Would it not be simpler to try to do something with re-invites or REFER, >>> assuming your endpoints support it? >> >> That was actually plan A. I already added a property in sip_profile called >> failover_redirect, which specifies another server to try if FS can't >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max >> Calls In Progress. > > You can't send a 302 to a call thats already established. Yes and I don't want to touch established calls - those calls can stay there until they drop. This is sent to new requests when switch_core_session_request fails in mod_sofia. >> Turns out not all my endpoints support it :( > > AKA broken endpoints. :) Some are broken. Some just have this feature disabled. For 'security reasons'. You know the drill. {P^/ John _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/56e045da/attachment-0002.html From stkn at freeswitch.org Fri Jun 12 10:21:18 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 12 Jun 2009 19:21:18 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A3266E5.2000702@ewetel.de> References: <4A3266E5.2000702@ewetel.de> Message-ID: <4A328E8E.6030607@freeswitch.org> Helmut Kuper wrote: > Hello, > > I just want to let you know, that I started some work on Q.931 TE state > timers in current openzap ozmod_isdn stack. The current stack has some > problems with freeing ressources when far end doesn't follow Q931 state > machine cleanly. On my side I mainly miss a RELEASE after sending > DISCONNECT and so m more and more channels are wasted over time. > > Openzap Q931 stack has some preworked timer and state handling (in form > of Q931_UX states). Unfortunately it is disabled in source. I enabled > it, added some Q931_UX states to Q931StateTE.c, added a timeout handler > to ozmod_isdn.c and tested it with my special problem. In lab missed > RELEASEs are now detected and the corresponding channel is freed cleanly > - hurray :) > > I would like to put my work in openzap trunk as soon as it works stable > in production and FS board allows me to do so. > > I can't promise that I will implement all timers for Q931 TE nor I plan > to work on Q931 NT timers. > > As I said: Just for your information. > > > Have a nice weekend! > > Helmut Umm, you've been doing duplicate work then. The version of ozmod_isdn i have been working on is completely stateful and has a couple of timers already implemented. And i remember giving you the location of the git repository on IRC, earlier this year. (But never got any feedback) stkn _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Fri Jun 12 10:52:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 10:52:09 -0700 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> Message-ID: <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> On Fri, Jun 12, 2009 at 9:26 AM, Andy wrote: > Excellent, thanks Anthony, I'll give it a go. > > Andy, can you report back on your success with this variable? Also, we would appreciate it if you could add an entry to the wiki on the channel_variables page. Let me know if you have any questions and I'll be glad to help. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/9e0a8ff3/attachment-0002.html From intralanman at freeswitch.org Fri Jun 12 11:10:33 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 12 Jun 2009 14:10:33 -0400 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C670262E23D62@mse17be1.mse17.exchange.ms><15A2DAAC-2EB5-4F26-A105-681A5E8B794F@avgs.ca><6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> Message-ID: <4A329A19.4010306@freeswitch.org> Saeed Ahmed wrote: > > No idea at all, > > It's a commercial SBC. > > I wish if we can have same functionality in FS. > You could accomplish parts of this with hearbeat and ldirectord.... the in-session calls aren't going to go anywhere, but if the server crashes, the second one can take over the ip of the first easily enough. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/2f9530a2/attachment-0002.html From msc at freeswitch.org Fri Jun 12 11:22:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 11:22:03 -0700 Subject: [Freeswitch-users] INFO: Important story you need to Digg and read right away Message-ID: <87f2f3b90906121122v6599ffbdxa061aab899ff15fe@mail.gmail.com> Gang, There have been crazy rumors flying around. Tony sets the record straight. Please go here now and digg this story: http://digg.com/software/Anthony_Minessale_of_FreeSWITCH_Discusses_Barracuda_Rumors Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/9b8b8742/attachment-0002.html From benj at teliax.com Fri Jun 12 11:59:14 2009 From: benj at teliax.com (Ben Jones) Date: Fri, 12 Jun 2009 12:59:14 -0600 Subject: [Freeswitch-users] RFC2833 double-digits Message-ID: <4A32A582.6040700@teliax.com> Hi all, We're running into a problem with rfc2833. Here's the situation: A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. The FS console shows it receives the digits once. We send the call to a tier-1 carrier The digits are played read back, but often the first several digits are repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when the user entered '1234.' When I was testing this recently, the call was sent to a Cisco gateway, which the documentation* states has problems with rfc2833, just like Sonus gateways, although that should be fixed past revision 10744. When using inband, everything works great. We are using start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers is inband. My initial thought was that the tier-1 was receiving rfc2833 and inband, but it's not consistent. I'm sure more information will be requested of me to help troubleshoot, so please let me know. Any advice would be very much appreciated. *http://wiki.freeswitch.org/wiki/RTP_Issues -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From brian at freeswitch.org Fri Jun 12 12:08:21 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 12 Jun 2009 14:08:21 -0500 Subject: [Freeswitch-users] RFC2833 double-digits In-Reply-To: <4A32A582.6040700@teliax.com> References: <4A32A582.6040700@teliax.com> Message-ID: <6B15EE47-6F85-4DC1-A3FB-928508D4E11C@freeswitch.org> What device are you using? RTP traces, debug logs something to see what might be taking place.?!?! /b On Jun 12, 2009, at 1:59 PM, Ben Jones wrote: > Hi all, > > We're running into a problem with rfc2833. Here's the situation: > > A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. > The FS console shows it receives the digits once. > We send the call to a tier-1 carrier > The digits are played read back, but often the first several digits > are > repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when > the user entered '1234.' > > When I was testing this recently, the call was sent to a Cisco > gateway, > which the documentation* states has problems with rfc2833, just like > Sonus gateways, although that should be fixed past revision 10744. > > When using inband, everything works great. We are using > start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers > is inband. My initial thought was that the tier-1 was receiving > rfc2833 > and inband, but it's not consistent. > > I'm sure more information will be requested of me to help > troubleshoot, > so please let me know. Any advice would be very much appreciated. > > *http://wiki.freeswitch.org/wiki/RTP_Issues Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/89a8231e/attachment-0002.html From benj at teliax.com Fri Jun 12 12:37:16 2009 From: benj at teliax.com (Ben Jones) Date: Fri, 12 Jun 2009 13:37:16 -0600 Subject: [Freeswitch-users] RFC2833 double-digits In-Reply-To: <6B15EE47-6F85-4DC1-A3FB-928508D4E11C@freeswitch.org> References: <4A32A582.6040700@teliax.com> <6B15EE47-6F85-4DC1-A3FB-928508D4E11C@freeswitch.org> Message-ID: <4A32AE6C.2030007@teliax.com> Testing was done with SJphone for Mac, dtmfmode rfc2833 pt 101. Hopefully this debug log can help: http://pastebin.freeswitch.org/9374 If I need to add, change, whatever, let me know. Thanks for the help. -benj Brian West wrote: > What device are you using? RTP traces, debug logs something to see what > might be taking place.?!?! > > /b > > On Jun 12, 2009, at 1:59 PM, Ben Jones wrote: > >> Hi all, >> >> We're running into a problem with rfc2833. Here's the situation: >> >> A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. >> The FS console shows it receives the digits once. >> We send the call to a tier-1 carrier >> The digits are played read back, but often the first several digits are >> repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when >> the user entered '1234.' >> >> When I was testing this recently, the call was sent to a Cisco gateway, >> which the documentation* states has problems with rfc2833, just like >> Sonus gateways, although that should be fixed past revision 10744. >> >> When using inband, everything works great. We are using >> start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers >> is inband. My initial thought was that the tier-1 was receiving rfc2833 >> and inband, but it's not consistent. >> >> I'm sure more information will be requested of me to help troubleshoot, >> so please let me know. Any advice would be very much appreciated. >> >> *http://wiki.freeswitch.org/wiki/RTP_Issues > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com From herb at powercom.com Fri Jun 12 15:09:10 2009 From: herb at powercom.com (Herb Levitin) Date: Fri, 12 Jun 2009 15:09:10 -0700 Subject: [Freeswitch-users] Need to hire an experienced FreeSwitch Developer Message-ID: <026701c9ebaa$6a8d90c0$3fa8b240$@com> I need to hire an experienced FreeSwitch developer to build a small chat bridge that supports VoIP and 1-4 PRI's of TDM. Please contact me at herb at powercom.com or (805)845-8906. From apt.get at gmail.com Fri Jun 12 15:14:42 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 12 Jun 2009 16:14:42 -0600 Subject: [Freeswitch-users] high latency Message-ID: Greetings, I'm new to freeswitch, just playing with it in a home/small office at present. Overall I've been really impressed with it. One issue we've noticed is a pronounced latency on some or perhaps all calls over roughly 500 seconds in duration. The latency is approximately 3 seconds or more round trip. I tried setting jitterbuffer_msec=20 but it has not prevented the problem. The call quality is otherwise good with no noticeable choppiness or jitter. Other local network traffic appears to be irrelevant, as in the latency seems to occur even when the LAN is idle or experiencing only sporadic bursts of activity, like typical browsing. I read the FAQ and searched jitter, lag and latency in the wiki and list archive but didn't come up with anything. Is this a known issue? Could it be that the problem is specific to my platform? I'm running the freeswitch package on pfsense (FreeBSD-based) and nobody in the pfsense forums seems to know the cause or solution, although others have reported the same issue. Much thanks for your input, db From msc at freeswitch.org Fri Jun 12 15:33:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 15:33:54 -0700 Subject: [Freeswitch-users] high latency In-Reply-To: References: Message-ID: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> Can you describe your setup? Need to know what kind of OS and hardware is running FS as well as what kind of phones. Any NAT involved? -MC On Fri, Jun 12, 2009 at 3:14 PM, David Burgess wrote: > Greetings, > > I'm new to freeswitch, just playing with it in a home/small office at > present. Overall I've been really impressed with it. > > One issue we've noticed is a pronounced latency on some or perhaps all > calls over roughly 500 seconds in duration. The latency is > approximately 3 seconds or more round trip. I tried setting > jitterbuffer_msec=20 but it has not prevented the problem. The call > quality is otherwise good with no noticeable choppiness or jitter. > Other local network traffic appears to be irrelevant, as in the > latency seems to occur even when the LAN is idle or experiencing only > sporadic bursts of activity, like typical browsing. > > I read the FAQ and searched jitter, lag and latency in the wiki and > list archive but didn't come up with anything. Is this a known issue? > Could it be that the problem is specific to my platform? I'm running > the freeswitch package on pfsense (FreeBSD-based) and nobody in the > pfsense forums seems to know the cause or solution, although others > have reported the same issue. > > Much thanks for your input, > > db > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/65872f1b/attachment-0002.html From apt.get at gmail.com Fri Jun 12 15:53:57 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 12 Jun 2009 16:53:57 -0600 Subject: [Freeswitch-users] high latency In-Reply-To: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> Message-ID: On Fri, Jun 12, 2009 at 4:33 PM, Michael Collins wrote: > Can you describe your setup? Need to know what kind of OS and hardware is > running FS as well as what kind of phones. Any NAT involved? > -MC FS is running inside pfsense, which is a freeBSD-based firewall (pfsense.org). Hardware is a lightly loaded Soekris net5501 (500 MHz Geode x86, 512 MB RAM)(www.soekris.com). FS external and internal profiles are both listening on WAN with a public IP address. My extensions are 2 lines on a Linksys PAP2T residing on the LAN, registered to the internal profile on the WAN interface. All calls are made via a sip trunk. PAP2T normally reports a decode latency in the neighborhood of 30 ms and jitter of 5 ms. Ping time to my provider's rtp servers is ~42 ms. Every call starts out with imperceptible latency, but at some point the caller notices a long delay, as I said, usually around 3 seconds, and usually only on calls lasting 500 seconds or more. Anything else I can provide? db From msc at freeswitch.org Fri Jun 12 16:28:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 16:28:15 -0700 Subject: [Freeswitch-users] high latency In-Reply-To: References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> Message-ID: <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> On Fri, Jun 12, 2009 at 3:53 PM, David Burgess wrote: > On Fri, Jun 12, 2009 at 4:33 PM, Michael Collins > wrote: > > Can you describe your setup? Need to know what kind of OS and hardware is > > running FS as well as what kind of phones. Any NAT involved? > > -MC > > FS is running inside pfsense, which is a freeBSD-based firewall > (pfsense.org). Hardware is a lightly loaded Soekris net5501 (500 MHz > Geode x86, 512 MB RAM)(www.soekris.com). > > FS external and internal profiles are both listening on WAN with a > public IP address. My extensions are 2 lines on a Linksys PAP2T > residing on the LAN, registered to the internal profile on the WAN > interface. All calls are made via a sip trunk. > > PAP2T normally reports a decode latency in the neighborhood of 30 ms > and jitter of 5 ms. Ping time to my provider's rtp servers is ~42 ms. > > Every call starts out with imperceptible latency, but at some point > the caller notices a long delay, as I said, usually around 3 seconds, > and usually only on calls lasting 500 seconds or more. > > Anything else I can provide? Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen anything like this with FS+pfSense. The FS devs aren't exactly keen on FS + FBSD because of some issues between a FS dependency (APR) and the FBSD threading model. Still, on a light load I wouldn't expect this kind of behavior. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/87d18253/attachment-0002.html From mrene_lists at avgs.ca Fri Jun 12 16:34:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 12 Jun 2009 19:34:52 -0400 Subject: [Freeswitch-users] high latency In-Reply-To: <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> Message-ID: Try Math On 12-Jun-09, at 7:28 PM, Michael Collins wrote: > > > On Fri, Jun 12, 2009 at 3:53 PM, David Burgess > wrote: > On Fri, Jun 12, 2009 at 4:33 PM, Michael Collins > wrote: > > Can you describe your setup? Need to know what kind of OS and > hardware is > > running FS as well as what kind of phones. Any NAT involved? > > -MC > > FS is running inside pfsense, which is a freeBSD-based firewall > (pfsense.org). Hardware is a lightly loaded Soekris net5501 (500 MHz > Geode x86, 512 MB RAM)(www.soekris.com). > > FS external and internal profiles are both listening on WAN with a > public IP address. My extensions are 2 lines on a Linksys PAP2T > residing on the LAN, registered to the internal profile on the WAN > interface. All calls are made via a sip trunk. > > PAP2T normally reports a decode latency in the neighborhood of 30 ms > and jitter of 5 ms. Ping time to my provider's rtp servers is ~42 ms. > > Every call starts out with imperceptible latency, but at some point > the caller notices a long delay, as I said, usually around 3 seconds, > and usually only on calls lasting 500 seconds or more. > > Anything else I can provide? > > Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen > anything like this with FS+pfSense. The FS devs aren't exactly keen > on FS + FBSD because of some issues between a FS dependency (APR) > and the FBSD threading model. Still, on a light load I wouldn't > expect this kind of behavior. > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/1d67d0f2/attachment-0002.html From john at feith.com Fri Jun 12 17:28:18 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 20:28:18 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130028.n5D0SIUF008628@jwlab.FEITH.COM> Upgraded from Apr 3 svn to svn 13769. Calling from openzap to 9999 (music on hold) works. Calling from openzap to 9995 (5 sec echo test) works. Calling from openzap to vmail works. Calling from Grandstream to 9999 (music on hold) works. Calling from Grandstream to 9995 (5 sec echo test) doesn't work ... call goes through however silence is heard. Calling from Grandstream to vmail doesn't work ... call goes through however vmail disconnects apparently due to receiving silence. Calling from Grandstream to openzap doesn't work ... call goes through and the Grandstream can hear what is said on the openzap side, however openzap hears silence from the Grandstream. Calling from Grandstream to Grandstream doesn't work ... call goes through however both sides hear silence. Suggestions on how to proceed? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From john at feith.com Fri Jun 12 17:48:21 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 20:48:21 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130048.n5D0mLAx008649@jwlab.FEITH.COM> BTW: in all cases show channels says PCMU 8000 is being used for the read and well as write codec. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From diego.viola at gmail.com Fri Jun 12 17:50:04 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 20:50:04 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH Message-ID: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> Hi everyone, I just found this project which seems to be a new VM for running languages... the parrot project aims to create a virtual machine for Perl 6 and other dynamic languages. You can take a look at it here: http://www.parrot.org/ It already supports many different languages: http://www.parrot.org/languages And it has a Apache module: http://www.parrot.org/mod_parrot I wonder that makes it embeddable... would it be possible to create a mod_parrot for FreeSWITCH? Please take this only as a feedback and not as anything else. Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/c318ed33/attachment-0002.html From diego.viola at gmail.com Fri Jun 12 17:53:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 20:53:05 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> Message-ID: <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> Here mercutioviz gave me some interesting info about Parrot. 20:47 <@mercutioviz> diegoviola: parrot isn't tied specifically to perl 6 but it is an offshoot of the perl 6 effort 20:47 <@mercutioviz> they were smart to break apart the big project into two separate projects 20:47 <@mercutioviz> parrot is strictly a virtual machine 20:48 <@mercutioviz> perl 6 is strictly a definition of a scripting language 20:48 <@mercutioviz> that lets people port all sorts of stuff to parrot without tripping over perl 6 20:49 <@mercutioviz> it also creates freakish possibilities, like calling perl library functions from a python script Regards, Diego On Fri, Jun 12, 2009 at 8:50 PM, Diego Viola wrote: > Hi everyone, > > I just found this project which seems to be a new VM for running > languages... the parrot project aims to create a virtual machine for Perl 6 > and other dynamic languages. You can take a look at it here: > > http://www.parrot.org/ > > It already supports many different languages: > > http://www.parrot.org/languages > > And it has a Apache module: > > http://www.parrot.org/mod_parrot > > I wonder that makes it embeddable... would it be possible to create a > mod_parrot for FreeSWITCH? > > Please take this only as a feedback and not as anything else. > > Diego > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/dee5f3b9/attachment-0002.html From msc at freeswitch.org Fri Jun 12 17:56:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Jun 2009 17:56:03 -0700 Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn In-Reply-To: <200906130028.n5D0SIUF008628@jwlab.FEITH.COM> References: <200906130028.n5D0SIUF008628@jwlab.FEITH.COM> Message-ID: <87f2f3b90906121756l4421b309k180f677f6a51b4a3@mail.gmail.com> On Fri, Jun 12, 2009 at 5:28 PM, John Wehle wrote: > Upgraded from Apr 3 svn to svn 13769. > That's a pretty decent jump. I think possibly that the configs changed, especially the auto-nat stuff. For kicks, try launching freeswitch with the "-nonat" flag and see if your symptoms persist. It may be that you need to get newer versions of the sip profile config files. If you didn't make any modifications to internal.xml and external.xml then delete those from conf/sip_profiles and then go to your fs source and do a "make samples" to get fresh copies of those two files. If you have modified those two files then I recommend looking at the new default config versions of those two files and integrating your changes into the new ones. Let us know how it goes... -MC > Calling from openzap to 9999 (music on hold) works. > Calling from openzap to 9995 (5 sec echo test) works. > Calling from openzap to vmail works. > > Calling from Grandstream to 9999 (music on hold) works. > Calling from Grandstream to 9995 (5 sec echo test) doesn't work > ... call goes through however silence is heard. > Calling from Grandstream to vmail doesn't work ... call goes > through however vmail disconnects apparently due to receiving silence. > > Calling from Grandstream to openzap doesn't work ... call goes > through and the Grandstream can hear what is said on the openzap > side, however openzap hears silence from the Grandstream. > > Calling from Grandstream to Grandstream doesn't work ... call goes > through however both sides hear silence. > > Suggestions on how to proceed? > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/45d58b19/attachment-0002.html From apt.get at gmail.com Fri Jun 12 17:58:34 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 12 Jun 2009 18:58:34 -0600 Subject: [Freeswitch-users] high latency In-Reply-To: References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> Message-ID: On Fri, Jun 12, 2009 at 5:34 PM, Mathieu Rene wrote: > Try Thanks, I will try that. > Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen anything like > this with FS+pfSense. Yeah, we've discussed it. He's experienced the same thing but doesn't have an answer at this point. The FS devs aren't exactly keen on FS + FBSD because > of some issues between a FS dependency (APR) and the FBSD threading model. > Still, on a light load I wouldn't expect this kind of behavior. Interesting. Thanks for the feedback. I'll let the list know what I find out. db From diego.viola at gmail.com Fri Jun 12 18:09:25 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 21:09:25 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> Message-ID: <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> They seem to have an embedding API here. http://docs.parrot.org/parrot/latest/html/docs/pdds/draft/pdd10_embedding.pod.html Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot support also. Diego On Fri, Jun 12, 2009 at 8:53 PM, Diego Viola wrote: > Here mercutioviz gave me some interesting info about Parrot. > > 20:47 <@mercutioviz> diegoviola: parrot isn't tied specifically to perl 6 > but it is an offshoot of the perl 6 effort > 20:47 <@mercutioviz> they were smart to break apart the big project into > two separate projects > 20:47 <@mercutioviz> parrot is strictly a virtual machine > 20:48 <@mercutioviz> perl 6 is strictly a definition of a scripting > language > 20:48 <@mercutioviz> that lets people port all sorts of stuff to parrot > without tripping over perl 6 > 20:49 <@mercutioviz> it also creates freakish possibilities, like calling > perl library functions from a python script > > Regards, > > Diego > > > On Fri, Jun 12, 2009 at 8:50 PM, Diego Viola wrote: > >> Hi everyone, >> >> I just found this project which seems to be a new VM for running >> languages... the parrot project aims to create a virtual machine for Perl 6 >> and other dynamic languages. You can take a look at it here: >> >> http://www.parrot.org/ >> >> It already supports many different languages: >> >> http://www.parrot.org/languages >> >> And it has a Apache module: >> >> http://www.parrot.org/mod_parrot >> >> I wonder that makes it embeddable... would it be possible to create a >> mod_parrot for FreeSWITCH? >> >> Please take this only as a feedback and not as anything else. >> >> Diego >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/6964502b/attachment-0002.html From jason at jasonjgw.net Fri Jun 12 18:15:07 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 13 Jun 2009 11:15:07 +1000 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> Message-ID: <20090613011507.GA13918@jdc.jasonjgw.net> Diego Viola wrote: > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot > support also. Are you offering to write it? From diego.viola at gmail.com Fri Jun 12 18:17:42 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 21:17:42 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <20090613011507.GA13918@jdc.jasonjgw.net> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> <20090613011507.GA13918@jdc.jasonjgw.net> Message-ID: <86a32abc0906121817n28a9ddcdu1d29b68b16aff66e@mail.gmail.com> No, I'm not a C programmer, just offering the idea (feedback). On Fri, Jun 12, 2009 at 9:15 PM, Jason White wrote: > Diego Viola wrote: > > > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot > > support also. > > Are you offering to write it? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/d8c27371/attachment-0002.html From john at feith.com Fri Jun 12 18:19:22 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 21:19:22 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130119.n5D1JMs0008730@jwlab.FEITH.COM> Yet more information ... a packet trace of a openzap to Grandstream call shows: Source Destination Packet FreeSWITCH Grandstream SIP Request: INVITE ... Grandstream FreeSWITCH SIP Status: 100 Trying Grandstream FreeSWITCH SIP Status: 180 Ringing Grandstream FreeSWITCH SIP Status: 200, with session description FreeSWITCH Grandstream SIP Request: ACK ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCH Grandstream SIP Request: BYE ... Grandstream FreeSWITCH SIP Status: 200 OK The interesting thing is I don't see the Grandstream attempt to send audio. Is there something that FreeSWITCH needs to say to the Grandstream in order for the phone to send audio? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From diego.viola at gmail.com Fri Jun 12 18:19:26 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 12 Jun 2009 21:19:26 -0400 Subject: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH In-Reply-To: <86a32abc0906121817n28a9ddcdu1d29b68b16aff66e@mail.gmail.com> References: <86a32abc0906121750g70ad49d3vd7d76bcb1cc7f3f1@mail.gmail.com> <86a32abc0906121753n6cfb5678ycc209f9cec12d459@mail.gmail.com> <86a32abc0906121809t66982ba8i8439215af6b7014@mail.gmail.com> <20090613011507.GA13918@jdc.jasonjgw.net> <86a32abc0906121817n28a9ddcdu1d29b68b16aff66e@mail.gmail.com> Message-ID: <86a32abc0906121819i6f9b5f27hc505cd145d738979@mail.gmail.com> Could I add a bounty? On Fri, Jun 12, 2009 at 9:17 PM, Diego Viola wrote: > No, I'm not a C programmer, just offering the idea (feedback). > > > On Fri, Jun 12, 2009 at 9:15 PM, Jason White wrote: > >> Diego Viola wrote: >> >> > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot >> > support also. >> >> Are you offering to write it? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090612/47f8739f/attachment-0002.html From john at feith.com Fri Jun 12 19:24:24 2009 From: john at feith.com (John Wehle) Date: Fri, 12 Jun 2009 22:24:24 -0400 (EDT) Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn Message-ID: <200906130224.n5D2OOC3008870@jwlab.FEITH.COM> > I think possibly that the configs changed, specially the auto-nat stuff Yep ... a closer look at the packet trace showed FreeSWITCH settings the Contact as 10.10.10.1 instead of the actual IP address of the machine. > If you have modified those two files then I recommend looking at the new > default config versions of those two files and integrating your changes into > the new ones. Yep ... that's my SOP. Looking at the internal.xml supplied with the new FS I see: Once I commented out those entries everything worked fine. I'm kind of surprised that this default changed ... the older FS came with these commented out and worked fine in the simple configuration where the server and phones are on the same network segment. In any case my config has been adjusted, things are working, it's Friday, and I get to go home so life is good. :-) -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From mcampbellsmith at gmail.com Sat Jun 13 01:44:00 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sat, 13 Jun 2009 18:44:00 +1000 Subject: [Freeswitch-users] Remove voicemail prompts Message-ID: <33c87fa30906130144s2ee1fc42n1c20b1ff2acfefb0@mail.gmail.com> Hi! How can I configure voicemail so that I do not get the options such as "record your message at the tone" and "mark this message as urgent" Thanks! From brian at freeswitch.org Sat Jun 13 09:14:40 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 12:14:40 -0400 Subject: [Freeswitch-users] No VoIP audio after upgrading to latest svn In-Reply-To: <200906130224.n5D2OOC3008870@jwlab.FEITH.COM> References: <200906130224.n5D2OOC3008870@jwlab.FEITH.COM> Message-ID: The bigger question is why was it finding that IP if it was wrong. /b On Jun 12, 2009, at 10:24 PM, John Wehle wrote: > Yep ... that's my SOP. > > Looking at the internal.xml supplied with the new FS I see: > > > > > Once I commented out those entries everything worked fine. > > I'm kind of surprised that this default changed ... the older FS came > with these commented out and worked fine in the simple configuration > where the server and phones are on the same network segment. > > In any case my config has been adjusted, things are working, it's > Friday, and I get to go home so life is good. :-) > > -- John Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/f1701780/attachment-0002.html From wiltingtree at gmail.com Sat Jun 13 09:18:54 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 13 Jun 2009 12:18:54 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly Message-ID: Hi, I have a question about mod_voicemail. I would like to use it independently of the dialplan and I was hoping to be able to add new voicemail accounts on-the-fly, without having to update the FreeSWITCH configuration files. But it seems to be forcing me to manually add each user to the dialplan. Does anybody know if mod_voicemail was made to work the way I'm trying to use it, and the best way to approach this? Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/d5d5b9e7/attachment-0002.html From Mailings at kh-dev.de Sat Jun 13 09:30:33 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sat, 13 Jun 2009 18:30:33 +0200 Subject: [Freeswitch-users] t38modem without registration... Message-ID: Hi all, maybe someone can help me here. For some compatibility issues I need to use an older version of t38modem which doesn't support SIP registration. Incoming faxes work well, but I don't know how to set this up for outgoing faxes. So, how can I setup this with FS for outgoing faxes without SIP registration? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/3a4fbec5/attachment-0002.html From brian at freeswitch.org Sat Jun 13 09:58:57 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 12:58:57 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: Why would have to touch the dialplan? Just one regex to catch it all.. and send it to voicemail. You still have to add users to the directory for them to have a mailbox anyway. /b On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > Hi, I have a question about mod_voicemail. I would like to use it > independently of the dialplan and I was hoping to be able to add new > voicemail accounts on-the-fly, without having to update the > FreeSWITCH configuration files. But it seems to be forcing me to > manually add each user to the dialplan. > > Does anybody know if mod_voicemail was made to work the way I'm > trying to use it, and the best way to approach this? > > Thanks, > Adam > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment-0002.html From wiltingtree at gmail.com Sat Jun 13 11:11:06 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 13 Jun 2009 14:11:06 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly Message-ID: I'm sorry, I said dialplan, but I was talking about the directory. So the directory file must be edited every time a new mailbox is created? > > Message: 7 > Date: Sat, 13 Jun 2009 12:58:57 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Why would have to touch the dialplan? Just one regex to catch it > all.. and send it to voicemail. You still have to add users to the > directory for them to have a mailbox anyway. > > /b > > On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > > > Hi, I have a question about mod_voicemail. I would like to use it > > independently of the dialplan and I was hoping to be able to add new > > voicemail accounts on-the-fly, without having to update the > > FreeSWITCH configuration files. But it seems to be forcing me to > > manually add each user to the dialplan. > > > > Does anybody know if mod_voicemail was made to work the way I'm > > trying to use it, and the best way to approach this? > > > > Thanks, > > Adam > > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 129 > ************************************************* > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/52ac1b3e/attachment-0002.html From krice at freeswitch.org Sat Jun 13 11:25:32 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 13 Jun 2009 13:25:32 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: Message-ID: Either that or you feed it from something else like XML_CURL From: Adam Wilt Reply-To: Date: Sat, 13 Jun 2009 14:11:06 -0400 To: Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly I'm sorry, I said dialplan, but I was talking about the directory.? So the directory file must be edited every time a new mailbox is created? ? > > Message: 7 > Date: Sat, 13 Jun 2009 12:58:57 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Why would have to touch the dialplan? ?Just one regex to catch it > all.. and send it to voicemail. ?You still have to add users to the > directory for them to have a mailbox anyway. > > /b > > On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > >> > Hi, I have a question about mod_voicemail. I would like to use it >> > independently of the dialplan and I was hoping to be able to add new >> > voicemail accounts on-the-fly, without having to update the >> > FreeSWITCH configuration files. But it seems to be forcing me to >> > manually add each user to the dialplan. >> > >> > Does anybody know if mod_voicemail was made to work the way I'm >> > trying to use it, and the best way to approach this? >> > >> > Thanks, >> > Adam >> > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/ba > d009e9/attachment.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 129 > ************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/7bd28f05/attachment-0002.html From sprice at gmail.com Sat Jun 13 11:31:26 2009 From: sprice at gmail.com (SP) Date: Sat, 13 Jun 2009 13:31:26 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: <7e2ac3270906131131w6ec5c80bvc29e98ca8c252b9c@mail.gmail.com> Unless you use mod_xml_curl On Sat, Jun 13, 2009 at 13:11, Adam Wilt wrote: > I'm sorry, I said dialplan, but I was talking about the directory. > So the directory file must be edited every time a new mailbox is created? > > > >> >> Message: 7 >> Date: Sat, 13 Jun 2009 12:58:57 -0400 >> From: Brian West >> Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset="us-ascii" >> >> >> Why would have to touch the dialplan? Just one regex to catch it >> all.. and send it to voicemail. You still have to add users to the >> directory for them to have a mailbox anyway. >> >> /b >> >> On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: >> >> > Hi, I have a question about mod_voicemail. I would like to use it >> > independently of the dialplan and I was hoping to be able to add new >> > voicemail accounts on-the-fly, without having to update the >> > FreeSWITCH configuration files. But it seems to be forcing me to >> > manually add each user to the dialplan. >> > >> > Does anybody know if mod_voicemail was made to work the way I'm >> > trying to use it, and the best way to approach this? >> > >> > Thanks, >> > Adam >> > >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> End of Freeswitch-users Digest, Vol 36, Issue 129 >> ************************************************* >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/dfb8a0aa/attachment-0002.html From christian.loeschenkohl at xpirio.com Sat Jun 13 10:57:52 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sat, 13 Jun 2009 19:57:52 +0200 Subject: [Freeswitch-users] mod_php needed Message-ID: <4A33E8A0.1070708@xpirio.com> hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian L?schenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From larclap at yahoo.com Sat Jun 13 12:42:15 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 13 Jun 2009 12:42:15 -0700 Subject: [Freeswitch-users] Error in Dialplan documentation? Message-ID: <006e01c9ec5f$0de046a0$29a0d3e0$@com> At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the top under From Dialplan, it says: Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml. However, I cannot find ${sip_profile} in freeswitch.xml. Is the documentation correct? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/b564c8ca/attachment-0002.html From brian at freeswitch.org Sat Jun 13 13:08:21 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 15:08:21 -0500 Subject: [Freeswitch-users] Error in Dialplan documentation? In-Reply-To: <006e01c9ec5f$0de046a0$29a0d3e0$@com> References: <006e01c9ec5f$0de046a0$29a0d3e0$@com> Message-ID: Yes look at the default dialplan... you should note that its in the default only. /b On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote: > At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, > near the top under From Dialplan, it says: > Bridge the incoming call to extension 100 and 101. The '%' is used > instead of the @ to indicate that the endpoints are registered > locally. Separate multiple endpoints with a comma. The $ > {sip_profile} variable is defined in freeswitch.xml. > > However, I cannot find ${sip_profile} in freeswitch.xml. Is the > documentation correct? > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/7690bdbe/attachment-0002.html From msc at freeswitch.org Sat Jun 13 13:56:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 13 Jun 2009 13:56:48 -0700 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <4A33E8A0.1070708@xpirio.com> References: <4A33E8A0.1070708@xpirio.com> Message-ID: <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian L?schenkohl > hello > > i am working for an austrian voip carrier. > for a few months i work with freeswitch and it is simply great. > it solves our needs in many places (high volume, flexible, stable). > the only thing i really miss is the avalibilty of php as a call control > language. > mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't > that good (or even there :-) ). > i know there is perl, i also implemented some applications (conference > system with provisioning, > inbound call routing to our application servers, some tests as pbx), but > what should i say - > perl and me aren't compatible in the end. > > the greatest thing for us would be that mod_php comes alive again with the > functional state > of mod_perl ( > http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > there is also a bounty entry about mod_php, to pay for this would also be > an option and > could be discussed. > > keep on with the excellent work and greetings from austria > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung& Entwicklung VoIP > > xpirio > Telekommunikation& Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/4c438e90/attachment-0002.html From larclap at yahoo.com Sat Jun 13 14:16:13 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 13 Jun 2009 14:16:13 -0700 Subject: [Freeswitch-users] Error in Dialplan documentation? In-Reply-To: References: <006e01c9ec5f$0de046a0$29a0d3e0$@com> Message-ID: <008601c9ec6c$2ea0c100$8be24300$@com> I'm sorry, but I can't find ${sip_profile} defined in any document in the conf directory. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, June 13, 2009 1:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error in Dialplan documentation? Yes look at the default dialplan... you should note that its in the default only. /b On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote: At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the top under From Dialplan, it says: Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml. However, I cannot find ${sip_profile} in freeswitch.xml. Is the documentation correct? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/b68fae43/attachment-0002.html From brian at freeswitch.org Sat Jun 13 14:21:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 16:21:30 -0500 Subject: [Freeswitch-users] Error in Dialplan documentation? In-Reply-To: <008601c9ec6c$2ea0c100$8be24300$@com> References: <006e01c9ec5f$0de046a0$29a0d3e0$@com> <008601c9ec6c$2ea0c100$8be24300$@com> Message-ID: <86F9DFF8-8A2C-49FF-9AD2-771529E9BE23@freeswitch.org> Its not there anymore... use_profile is. But its just a variable so leaving it as is .. its prob. the best.. /b On Jun 13, 2009, at 4:16 PM, Lars Zeb wrote: > I?m sorry, but I can?t find ${sip_profile} defined in any document > in the conf directory. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Saturday, June 13, 2009 1:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error in Dialplan documentation? > > Yes look at the default dialplan... you should note that its in the > default only. > > /b > > On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote: > > > At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, > near the top under From Dialplan, it says: > Bridge the incoming call to extension 100 and 101. The '%' is used > instead of the @ to indicate that the endpoints are registered > locally. Separate multiple endpoints with a comma. The $ > {sip_profile} variable is defined in freeswitch.xml. > > However, I cannot find ${sip_profile} in freeswitch.xml. Is the > documentation correct? > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bbcd6c82/attachment-0002.html From wiltingtree at gmail.com Sat Jun 13 15:28:55 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 13 Jun 2009 18:28:55 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly Message-ID: Thanks. I would really like mod_voicemail to be database driven, instead of by XML and cURL. I noticed in the documentation that you can provide an ODBC handle in the voicemail.conf.xml file, and so I tried it with MySQL. It created the two tables, voicemail_msgs and voicemail_prefs properly, and voicemail_msgs works the way I expect; when a new voicemail is created it writes a new record in this table. But I don't understand the purpose of voicemail_prefs; when I add a record here with a username and password, mod_voicemail ignores it. So I still have to use the config file or xml_curl to set-up the users. It doesn't seem like mod_voicemail reads from or writes to this table. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/83729b49/attachment-0002.html From krice at freeswitch.org Sat Jun 13 15:41:38 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 13 Jun 2009 17:41:38 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: Message-ID: That does not provide for User configuration only voicemail metadata storage... Again, at this time you would need to use XML CURL to do what you are trying to do. There at some point in the future may be a way to grab users via direct ODBC or other database driver however at this point its not possible... Your only other possible option is via LDAP for the users From: Adam Wilt Reply-To: Date: Sat, 13 Jun 2009 18:28:55 -0400 To: Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly Thanks. I?would really like?mod_voicemail to be database driven, instead of by XML and cURL. I noticed in the documentation that you can provide an ODBC handle in the voicemail.conf.xml file, and so I tried it with MySQL. It created the two tables, voicemail_msgs and voicemail_prefs properly, and voicemail_msgs works the way I expect; when a new voicemail is created it writes a new record in this table.? But I don't understand the purpose of voicemail_prefs;? when I add a record here with a username and password, mod_voicemail ignores it. So I still have to use the config file or xml_curl to set-up the users. It doesn't seem like mod_voicemail reads from or writes to this table. ? ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/553eeec1/attachment-0002.html From brian at freeswitch.org Sat Jun 13 15:42:12 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 13 Jun 2009 17:42:12 -0500 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: Then you're free to write your own directory hook and plug it directly into the database how ever you wish. Look how XML_CURL uses this interface. /b On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote: > Thanks. I would really like mod_voicemail to be database driven, > instead of by XML and cURL. > I noticed in the documentation that you can provide an ODBC handle > in the voicemail.conf.xml file, and so I tried it with MySQL. > It created the two tables, voicemail_msgs and voicemail_prefs > properly, and voicemail_msgs works the way I expect; when a new > voicemail is created it writes a new record in this table. But I > don't understand the purpose of voicemail_prefs; when I add a > record here with a username and password, mod_voicemail ignores it. > So I still have to use the config file or xml_curl to set-up the > users. It doesn't seem like mod_voicemail reads from or writes to > this table. From diego.viola at gmail.com Sat Jun 13 16:07:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 13 Jun 2009 19:07:02 -0400 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: <86a32abc0906131607k6c67dc8ay95ad1f7948498317@mail.gmail.com> And please contribute things back when you do it. Diego On Sat, Jun 13, 2009 at 6:42 PM, Brian West wrote: > Then you're free to write your own directory hook and plug it directly > into the database how ever you wish. Look how XML_CURL uses this > interface. > > /b > > On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote: > > > Thanks. I would really like mod_voicemail to be database driven, > > instead of by XML and cURL. > > I noticed in the documentation that you can provide an ODBC handle > > in the voicemail.conf.xml file, and so I tried it with MySQL. > > It created the two tables, voicemail_msgs and voicemail_prefs > > properly, and voicemail_msgs works the way I expect; when a new > > voicemail is created it writes a new record in this table. But I > > don't understand the purpose of voicemail_prefs; when I add a > > record here with a username and password, mod_voicemail ignores it. > > So I still have to use the config file or xml_curl to set-up the > > users. It doesn't seem like mod_voicemail reads from or writes to > > this table. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/9033e215/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Jun 13 16:15:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 14 Jun 2009 00:15:53 +0100 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> Message-ID: I couldn't agree more. We're working with a group that are developing a massive PHP based music application. They are experts in PHP and MySQL but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate with the FS event socket, allows them to work on the areas they know best and not worry about the telephony side too much. We went the lua route, and don't use the dial plan at all. My view is to keep all db access and processing out of FS as much as possible. With the event socket you simply don't need to embed anything apart from the essentials. We are now processing 100,000+ call setups a day (4 hours) per server all using php scripts to drive the application. We may well ultimately use C++ instead of PHP for the event socket comms, but right now PHP does just fine. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 13 June 2009 21:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_php needed Perhaps you should look at controlling calls via the FreeSWITCH event socket instead of from the dialplan. The nice thing about the event socket is that your call control can happen on a separate machine. There is a PHP module for the ESL (event socket library) and it would be relatively easy for you to get going. Here are some links to get you started: http://wiki.freeswitch.org/wiki/PHP_Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket If you absolutely MUST have call control with scripts inside of the dialplan then there simply is no better choice than Lua. You can learn Lua in a few hours, but getting mod_php finished and debugged will take time, money, and other resources that no one seems willing to spend. Here is some information to consider: http://wiki.freeswitch.org/wiki/Mod_lua Come join us on IRC (#freeswitch on irc.freenode.net) if you want to discuss this further. -MC (IRC: mercutioviz) 2009/6/13 Christian L?schenkohl hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian L?schenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090614/2cd4e41f/attachment-0002.html From darklion11 at yahoo.com Sat Jun 13 23:11:43 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sat, 13 Jun 2009 23:11:43 -0700 (PDT) Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? Message-ID: <24018568.post@talk.nabble.com> Ip Windows: 192.168.0.104 Ip Linux: 192.168.0.105 My windows: My sample on sip_profiles/external/dialus.xml My linux: My sample on sip_profiles/external/dialus2.xml I have a number on windows 01497710001, on linux 01497710002 Trying to call each other on windows I dial 0149771002 But error on switch_ivr_originate: INVALID_NUMBER_FORMAT Please help me with this urgent issue... Or send me instructions or xml code that will me to solve this issue... Thanks, Edmar -- View this message in context: http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sun Jun 14 09:39:15 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 14 Jun 2009 11:39:15 -0500 Subject: [Freeswitch-users] MPL Confusion In-Reply-To: References: Message-ID: <950BE21A-5B01-4983-8AF3-1C327EEC5B7D@freeswitch.org> For clarification ... Read section 3.2 and 3.3 of the MPL 1.1 The simplest way I can describe it is how it was described to me "What's yours is yours and what's mine is mine!". /b On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: > I have some confusion about FreeSWITCH's Mozilla Public License 1.1. > I do understand that me or any one can change provided code > according to our customization needs and we are not bound to share > our changes as long as we are not distributing it, right? From gavin.henry at gmail.com Sun Jun 14 13:14:48 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Sun, 14 Jun 2009 21:14:48 +0100 Subject: [Freeswitch-users] mod_voicemail accounts on-the-fly In-Reply-To: References: Message-ID: <13ca621c0906141314sc9abde9p958fec8cb4772491@mail.gmail.com> Are there any ldap examples? On 13/06/2009, Ken Rice wrote: > That does not provide for User configuration only voicemail metadata > storage... Again, at this time you would need to use XML CURL to do what you > are trying to do. There at some point in the future may be a way to grab > users via direct ODBC or other database driver however at this point its not > possible... Your only other possible option is via LDAP for the users > > > > From: Adam Wilt > Reply-To: > Date: Sat, 13 Jun 2009 18:28:55 -0400 > To: > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > > Thanks. I?would really like?mod_voicemail to be database driven, instead of > by XML and cURL. > I noticed in the documentation that you can provide an ODBC handle in the > voicemail.conf.xml file, and so I tried it with MySQL. > It created the two tables, voicemail_msgs and voicemail_prefs properly, and > voicemail_msgs works the way I expect; when a new voicemail is created it > writes a new record in this table.? But I don't understand the purpose of > voicemail_prefs;? when I add a record here with a username and password, > mod_voicemail ignores it. So I still have to use the config file or xml_curl > to set-up the users. It doesn't seem like mod_voicemail reads from or writes > to this table. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From gavin.henry at gmail.com Sun Jun 14 13:34:15 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Sun, 14 Jun 2009 21:34:15 +0100 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? Message-ID: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From nik.middleton at noblesolutions.co.uk Sun Jun 14 14:38:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 14 Jun 2009 22:38:02 +0100 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? In-Reply-To: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> References: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> Message-ID: Anything that's dedicated undoubtedly has less load that something that's multifunctioned. However the lack of any conversations on front ending a SIP server to FS would likely indicate that no one's found a requirement for it at this time. I would truly hate to see discussions of theoretical performance advantages of one SIP server over another, when in my view, I have yet to reach any real world limit with FS. My FS servers are handling 100,000+ calls/day per server and are probably only at 50% capacity. (I see no point in beating a server to pulp when it's relatively cheap to add another if required) Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gavin Henry Sent: 14 June 2009 21:34 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From drago at windstream.net Sun Jun 14 16:11:27 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 14 Jun 2009 19:11:27 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Message-ID: <000001c9ed45$72058320$56108960$@net> Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: "MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The 'From' field's address to be in the format ''; otherwise, MS Exchange drops the call." I don't know if this is the only problem. However, I see exactly this behavior: "PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established." After "302 (Moved Temporarily", FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090614/ecf43fe4/attachment-0002.html From jingwei.yang at gmail.com Sun Jun 14 19:29:24 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 15 Jun 2009 10:29:24 +0800 Subject: [Freeswitch-users] Possible to initiate multiple skype interfaces with the same id? Message-ID: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> Hi Team, As the subject indicates, is there a possible way to do that? I've tried setting up two different skype instances with the same id in /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh /usr/bin/Xvfb :101 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype --pipelogin &" /usr/bin/Xvfb :102 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype --pipelogin &" The script can be run without any error and when I executed the ps command, I noticed there's only one skype process up. This is strange since normally there will be two (with two different ids) 3782 pts/1 00:00:00 Xvfb 3793 pts/1 00:00:00 dbus-launch 3800 pts/1 00:00:04 Xvfb 3805 pts/1 00:00:03 *skype* 3811 pts/1 00:00:00 dbus-launch Then I started freeswitch and hit an error when loading mod_skypiax: *freeswitch at localhost.localdomain> load mod_skypiax 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING interface_id=1 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error code 3 from X Server 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending message failed with status 3 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error code 3 from X Server 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending message failed with status 3 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==userAAA API CALL [load(mod_skypiax)] output: -ERR [module load file routine returned an error] 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' has been found. Please (re)launch a Skype client logged in as 'userAAA'. Skypiax exiting now 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_skypiax.so **Module load routine returned an error** * Please enlighten whether it's possible to start multiple skype instances with the same skype id. If possible, what are the correct configs? Thanks, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/c438c7cb/attachment-0002.html From msc at freeswitch.org Sun Jun 14 19:51:38 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 14 Jun 2009 19:51:38 -0700 Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? In-Reply-To: <24018568.post@talk.nabble.com> References: <24018568.post@talk.nabble.com> Message-ID: <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> Just curious, why are you dialing out the external gw? -MC Sent from my iPhone On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote: > > Ip Windows: 192.168.0.104 > Ip Linux: 192.168.0.105 > > > My windows: > > My sample on sip_profiles/external/dialus.xml > > > > > > data="sofia/external/$1 at 192.168.0.105:5080"/> > > > > > > My linux: > > My sample on sip_profiles/external/dialus2.xml > > > > > > data="sofia/external/$1 at 192.168.0.104:5080"/> > > > > > > I have a number on windows 01497710001, on linux 01497710002 > > Trying to call each other on windows I dial 0149771002 > > But error on switch_ivr_originate: INVALID_NUMBER_FORMAT > > Please help me with this urgent issue... > > Or send me instructions or xml code that will me to solve this > issue... > > > Thanks, > > Edmar > -- > View this message in context: http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul.degt at gmail.com Sun Jun 14 20:59:38 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Sun, 14 Jun 2009 23:59:38 -0400 Subject: [Freeswitch-users] session.getDigits() not working Message-ID: <4A35C72A.6030804@gmail.com> Trying out latest trunk ans seems like js function session.getDigits() stopped working (not collecting any digits), I do see switch_rtp.c:1560 Send end packet for [5] ts=2222260 dur=2080/2080/2000 seq=8732 in debug log so I assume dtmf is ok. Anybody can shed some light on why wouldn't it work now? Works just fine under 1.0.3 release. I use slightly modified version of disa.js from fs examples. Thanks. From shaheryarkh at googlemail.com Sun Jun 14 21:07:23 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 15 Jun 2009 10:07:23 +0600 Subject: [Freeswitch-users] MPL Confusion In-Reply-To: <950BE21A-5B01-4983-8AF3-1C327EEC5B7D@freeswitch.org> References: <950BE21A-5B01-4983-8AF3-1C327EEC5B7D@freeswitch.org> Message-ID: Thanks, I will look at it in more details as you suggested. I try to be online to discuss mod_msn and mod_yahoo on FS IRC channel this after noon Danish time. Thank you. On Sun, Jun 14, 2009 at 10:39 PM, Brian West wrote: > For clarification ... Read section 3.2 and 3.3 of the MPL 1.1 > > The simplest way I can describe it is how it was described to me > "What's yours is yours and what's mine is mine!". > > /b > > On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: > > > I have some confusion about FreeSWITCH's Mozilla Public License 1.1. > > I do understand that me or any one can change provided code > > according to our customization needs and we are not bound to share > > our changes as long as we are not distributing it, right? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/260babd8/attachment-0002.html From shaheryarkh at googlemail.com Sun Jun 14 21:21:50 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 15 Jun 2009 10:21:50 +0600 Subject: [Freeswitch-users] Possible to initiate multiple skype interfaces with the same id? In-Reply-To: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> References: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> Message-ID: I don't think you can start more then one Skype instance of same Skype ID from a single machine. You can either login with same Skype ID on two (or more different machines) or use two (or more) different Skype IDs on same machine. Thank you. On Mon, Jun 15, 2009 at 8:29 AM, Jingwei Yang wrote: > Hi Team, > > As the subject indicates, is there a possible way to do that? > > I've tried setting up two different skype instances with the same id in > /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh > > /usr/bin/Xvfb :101 -auth > /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & > su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype > --pipelogin &" > > /usr/bin/Xvfb :102 -auth > /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & > su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype > --pipelogin &" > > The script can be run without any error and when I executed the ps command, > I noticed there's only one skype process up. This is strange since normally > there will be two (with two different ids) > > 3782 pts/1 00:00:00 Xvfb > 3793 pts/1 00:00:00 dbus-launch > 3800 pts/1 00:00:04 Xvfb > 3805 pts/1 00:00:03 *skype* > 3811 pts/1 00:00:00 dbus-launch > > Then I started freeswitch and hit an error when loading mod_skypiax: > > *freeswitch at localhost.localdomain> load mod_skypiax > 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev > 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING > interface_id=1 > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev > 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error > code 3 from X Server > > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() > rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending > message failed with status 3 > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev > 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error > code 3 from X Server > > 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() > rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending > message failed with status 3 > 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev > 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 > seconds to find a running Skype client and connect to its SKYPE API for > interface_id=1 > 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev > 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running > Skype client, connected to its SKYPE API for interface_id=1, waiting 60 > seconds for CURRENTUSERHANDLE==userAAA > API CALL [load(mod_skypiax)] output: > -ERR [module load file routine returned an error] > > 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev > 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype client > to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, > interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' has > been found. Please (re)launch a Skype client logged in as 'userAAA'. > Skypiax exiting now > 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_skypiax.so > **Module load routine returned an error** > > * > Please enlighten whether it's possible to start multiple skype instances > with the same skype id. If possible, what are the correct configs? > > Thanks, > -Jingwei > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/b1651808/attachment-0002.html From jingwei.yang at gmail.com Sun Jun 14 22:12:02 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 15 Jun 2009 13:12:02 +0800 Subject: [Freeswitch-users] Possible to initiate multiple skype interfaces with the same id? In-Reply-To: References: <13529f9d0906141929q93adc2dy473a0a214af1a5df@mail.gmail.com> Message-ID: <13529f9d0906142212m690d4176xa735f54886ee63e8@mail.gmail.com> Hi Muhammad, thanks for the reply. On Mon, Jun 15, 2009 at 12:21 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I don't think you can start more then one Skype instance of same Skype ID > from a single machine. You can either login with same Skype ID on two (or > more different machines) or use two (or more) different Skype IDs on same > machine. > > Thank you. > > > On Mon, Jun 15, 2009 at 8:29 AM, Jingwei Yang wrote: > >> Hi Team, >> >> As the subject indicates, is there a possible way to do that? >> >> I've tried setting up two different skype instances with the same id in >> /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh >> >> /usr/bin/Xvfb :101 -auth >> /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & >> su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype >> --pipelogin &" >> >> /usr/bin/Xvfb :102 -auth >> /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & >> su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype >> --pipelogin &" >> >> The script can be run without any error and when I executed the ps >> command, I noticed there's only one skype process up. This is strange since >> normally there will be two (with two different ids) >> >> 3782 pts/1 00:00:00 Xvfb >> 3793 pts/1 00:00:00 dbus-launch >> 3800 pts/1 00:00:04 Xvfb >> 3805 pts/1 00:00:03 *skype* >> 3811 pts/1 00:00:00 dbus-launch >> >> Then I started freeswitch and hit an error when loading mod_skypiax: >> >> *freeswitch at localhost.localdomain> load mod_skypiax >> 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev >> 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING >> interface_id=1 >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev >> 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error >> code 3 from X Server >> >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() >> rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending >> message failed with status 3 >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev >> 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error >> code 3 from X Server >> >> 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() >> rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending >> message failed with status 3 >> 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev >> 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 >> seconds to find a running Skype client and connect to its SKYPE API for >> interface_id=1 >> 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev >> 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >> seconds for CURRENTUSERHANDLE==userAAA >> API CALL [load(mod_skypiax)] output: >> -ERR [module load file routine returned an error] >> >> 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev >> 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype >> client to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, >> interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' >> has been found. Please (re)launch a Skype client logged in as 'userAAA'. >> Skypiax exiting now >> 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 >> switch_loadable_module_load_file() Error Loading module >> /usr/local/freeswitch/mod/mod_skypiax.so >> **Module load routine returned an error** >> >> * >> Please enlighten whether it's possible to start multiple skype instances >> with the same skype id. If possible, what are the correct configs? >> >> Thanks, >> -Jingwei >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/3312ad0c/attachment-0002.html From darklion11 at yahoo.com Sun Jun 14 22:24:05 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 14 Jun 2009 22:24:05 -0700 (PDT) Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? In-Reply-To: <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> References: <24018568.post@talk.nabble.com> <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> Message-ID: <1245043445817-3078668.post@n2.nabble.com> Actually, I dont know if my xml code is correct.. Please give me an example of an external profile connecting this 2 OS? And instructions Thanks mercutioviz wrote: > > Just curious, why are you dialing out the external gw? > -MC > > Sent from my iPhone > > On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote: > >> >> Ip Windows: 192.168.0.104 >> Ip Linux: 192.168.0.105 >> >> >> My windows: >> >> My sample on sip_profiles/external/dialus.xml >> >> >> >> >> >> data="sofia/external/$1 at 192.168.0.105:5080"/> >> >> >> >> >> >> My linux: >> >> My sample on sip_profiles/external/dialus2.xml >> >> >> >> >> >> data="sofia/external/$1 at 192.168.0.104:5080"/> >> >> >> >> >> >> I have a number on windows 01497710001, on linux 01497710002 >> >> Trying to call each other on windows I dial 0149771002 >> >> But error on switch_ivr_originate: INVALID_NUMBER_FORMAT >> >> Please help me with this urgent issue... >> >> Or send me instructions or xml code that will me to solve this >> issue... >> >> >> Thanks, >> >> Edmar >> -- >> View this message in context: >> http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp3074838p3078668.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090614/366e2f16/attachment-0002.html From darklion11 at yahoo.com Sun Jun 14 22:27:27 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 14 Jun 2009 22:27:27 -0700 (PDT) Subject: [Freeswitch-users] How to connect Freeswitch Windows, Freeswitch Linux and also vice versa? In-Reply-To: <1245043445817-3078668.post@n2.nabble.com> References: <24018568.post@talk.nabble.com> <4468FE05-8EED-4256-BEA4-26B455F1BA4F@freeswitch.org> <1245043445817-3078668.post@n2.nabble.com> Message-ID: <1245043647184-3078676.post@n2.nabble.com> I just want to connect to this to OS with an external xml code. How can I do that? I get some examples on the wiki but not working. Edmar Cruz wrote: > > Actually, I dont know if my xml code is correct.. Please give me an > example of an external profile connecting this 2 OS? And instructions > > Thanks > > > > mercutioviz wrote: >> >> Just curious, why are you dialing out the external gw? >> -MC >> >> Sent from my iPhone >> >> On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote: >> >>> >>> Ip Windows: 192.168.0.104 >>> Ip Linux: 192.168.0.105 >>> >>> >>> My windows: >>> >>> My sample on sip_profiles/external/dialus.xml >>> >>> >>> >>> >>> >>> >>> data="sofia/external/$1 at 192.168.0.105:5080"/> >>> >>> >>> >>> >>> >>> My linux: >>> >>> My sample on sip_profiles/external/dialus2.xml >>> >>> >>> >>> >>> >>> >>> data="sofia/external/$1 at 192.168.0.104:5080"/> >>> >>> >>> >>> >>> >>> I have a number on windows 01497710001, on linux 01497710002 >>> >>> Trying to call each other on windows I dial 0149771002 >>> >>> But error on switch_ivr_originate: INVALID_NUMBER_FORMAT >>> >>> Please help me with this urgent issue... >>> >>> Or send me instructions or xml code that will me to solve this >>> issue... >>> >>> >>> Thanks, >>> >>> Edmar >>> -- >>> View this message in context: >>> http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp24018568p24018568.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux-and-also-vice-versa--tp3074838p3078676.html Sent from the freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Sun Jun 14 23:57:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 15 Jun 2009 08:57:52 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A328E8E.6030607@freeswitch.org> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> Message-ID: <4A35F0F0.50406@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Stefan, On 12.06.2009 19:21, Stefan Knoblich wrote: > Umm, you've been doing duplicate work then. :( Well, I implemented just one timer by now. So not much time has been wasted ... > > The version of ozmod_isdn i have been working on is completely stateful and > has a couple of timers already implemented. Very good :) > > And i remember giving you the location of the git repository on IRC, > earlier this year. (But never got any feedback) Sorry, at that time we talked about q931 to pcap. By now I thought state timers are still not done, so there wasn't a reason to test it regarding state timers .... buuuut today things changed and I will download your openzap. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKNfDw4tZeNddg3dwRAkThAJ4iPiZ4ZOAKKWpmKdCbjM8oM5mH6QCeMo5m pV4Y1/hpO7osV8cuInYJd2o= =zoZu -----END PGP SIGNATURE----- From dujinfang at gmail.com Mon Jun 15 00:41:49 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 15:41:49 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls Message-ID: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> Hi, I'm on version 13524, call from zoiper is ok, but when call zoiper, it keep rejecting calls, anyone can help? I'm seems always not the right time join in IRC :( http://pastebin.freeswitch.org/9383 Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1a55a71a/attachment-0002.html From Claudio.Cavalera at italtel.it Mon Jun 15 01:30:16 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 15 Jun 2009 10:30:16 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <74170625-2AA0-4FF6-969D-3C850DCD7CA0@jerris.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: >> OTOH there will be a bit of trouble getting the internal state out >> of all those modules and libraries... in particular sofia :D > > We have talked quite some about this, its a major job, easily months > of work for multiple programmers. We would love to do it but > its not > on any roadmaps at this time. > Could this be also achieved in hardware via ATCA ? en.wikipedia.org/wiki/Advanced_Telecommunications_Computing_Architecture Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From gavin.henry at gmail.com Mon Jun 15 01:47:54 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 15 Jun 2009 09:47:54 +0100 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? In-Reply-To: References: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> Message-ID: <13ca621c0906150147t504a31feof75a37d8f478a1f3@mail.gmail.com> OK, thanks. 2009/6/14 Nik Middleton : > Anything that's dedicated undoubtedly has less load that something > that's multifunctioned. ?However the lack of any conversations on front > ending a SIP server to FS would likely indicate that no one's found a > requirement for it at this time. > > I would truly hate to see discussions of theoretical performance > advantages of one SIP server over another, when in my view, I have yet > to reach any real world limit with FS. ?My FS servers are handling > 100,000+ calls/day per server and are probably only at 50% capacity. (I > see no point in beating a server to pulp when it's relatively cheap to > add another if required) > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Gavin Henry > Sent: 14 June 2009 21:34 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of > FS? > > Hi, > > I'm excited reading all the threads about how FS blows Asterisk away > so that you don't need ?OpenSIPS/Kamailio in front of FS. Surely there > must be a point when it would be advisable to do that though, as > mod_sofia can't be as good as a dedicated SIP proxy? > > Thanks. > > -- > Sent from my mobile device > > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From grevenx at me.com Mon Jun 15 02:01:27 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Mon, 15 Jun 2009 11:01:27 +0200 Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of FS? In-Reply-To: <13ca621c0906150147t504a31feof75a37d8f478a1f3@mail.gmail.com> References: <13ca621c0906141334j15c8f62fs24f236afddbfbec2@mail.gmail.com> <13ca621c0906150147t504a31feof75a37d8f478a1f3@mail.gmail.com> Message-ID: Well, we currently have a scenario where this seems to be the most logical setup currently. We provide PBX as a Service (SaaS), and want to have a cluster of FreeSwitch servers handling registration and presence. Introducing OpenSIPS in front will allow a couple of features, which I don't see how would be implemented in a "good" way without anything in front of FS: - "true" loadbalancing with the loadbalancer module - Live migration of calls to another server to take FS down for maintenance - no need for 100% SRV support in the SIP clients Best regards, Even Andr? On 15. juni. 2009, at 10.47, Gavin Henry wrote: > OK, thanks. > > > 2009/6/14 Nik Middleton : >> Anything that's dedicated undoubtedly has less load that something >> that's multifunctioned. However the lack of any conversations on >> front >> ending a SIP server to FS would likely indicate that no one's found a >> requirement for it at this time. >> >> I would truly hate to see discussions of theoretical performance >> advantages of one SIP server over another, when in my view, I have >> yet >> to reach any real world limit with FS. My FS servers are handling >> 100,000+ calls/day per server and are probably only at 50% >> capacity. (I >> see no point in beating a server to pulp when it's relatively cheap >> to >> add another if required) >> >> Regards >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Gavin Henry >> Sent: 14 June 2009 21:34 >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] When to use OpenSIPS/Kamailio in front of >> FS? >> >> Hi, >> >> I'm excited reading all the threads about how FS blows Asterisk away >> so that you don't need OpenSIPS/Kamailio in front of FS. Surely >> there >> must be a point when it would be advisable to do that though, as >> mod_sofia can't be as good as a dedicated SIP proxy? >> >> Thanks. >> >> -- >> Sent from my mobile device >> >> http://www.suretecsystems.com/services/openldap/ >> http://www.suretectelecom.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Mon Jun 15 02:33:07 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 15 Jun 2009 12:33:07 +0300 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Message-ID: <1245058387.4694.5.camel@dk-d820> On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote: > What is the current status of Freeswitch? Can I safely use it in a > large scale commercial environment? How active is the Freeswitch > developer community? Hi Paul - We've used FS over the last 18 months or so to handle millions of calls - some wholesale in/out, some IVR, some calling card, some callthrough - with a total value in the millions of dollars; we have no complaints. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From jingwei.yang at gmail.com Mon Jun 15 02:40:08 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 15 Jun 2009 17:40:08 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session Message-ID: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: And here's how I trigger it: *freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA 2909/userBBB* The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: *freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH->2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) * Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. Thanks! -Jingwei ** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/4b6d83ec/attachment-0002.html From christian.loeschenkohl at xpirio.com Mon Jun 15 02:40:24 2009 From: christian.loeschenkohl at xpirio.com (=?windows-1252?Q?Christian_L=F6schenkohl?=) Date: Mon, 15 Jun 2009 11:40:24 +0200 Subject: [Freeswitch-users] mod_php needed In-Reply-To: References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> Message-ID: <4A361708.8020808@xpirio.com> hi thank you very much for your input i can say for me that i realy tried hard to use the event socket library, but untill now i can't use it like i used all the agi scripts or even mod_perl now. what i do most - in examples, if the server get's an incomming call - find the right user for the number (not that easy because of did in austria), from database or file - build the right dial string for the bridge application (here i miss all the php string functions most) - unsing mod_php functions like setVariable, getVariable, answer, transfer, sleep (i don't see how to do this with the php esl) - or i check if the number is part of a conferencing product and build the right conference setup i think this would also be possible with lua and luasql, but i developed years with phpagi und i'm very used to php in every kind of scripting or how-to-get-a-solution situation (since over 10 years now). for me in our setup it's also the highest goal to get the servers mostly independent of each other. i think nobody of our costumers should be unreachable because a central scripting/event server or also database server has gone away (as developers this happens more often as we would like it to :-)) do not get me wrong, freeswitch is very powerfull and in the near future it will replace nearly all of our asterisk servers. in combination with php the freeswitch plattform would be heaven for me i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php), in combination of the mod_python rewrite (page was last modified in june 2007). br On 2009-06-14 01:15, Nik Middleton wrote: > I couldn?t agree more. We?re working with a group that are developing a > massive PHP based music application. They are experts in PHP and MySQL > but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP > to communicate with the FS event socket, allows them to work on the > areas they know best and not worry about the telephony side too much. > We went the lua route, and don?t use the dial plan at all. My view is > to keep all db access and processing out of FS as much as possible. With > the event socket you simply don?t need to embed anything apart from the > essentials. > > We are now processing 100,000+ call setups a day (4 hours) per server > all using php scripts to drive the application. We may well ultimately > use C++ instead of PHP for the event socket comms, but right now PHP > does just fine. > > Regards > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Michael Collins > *Sent:* 13 June 2009 21:57 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_php needed > > Perhaps you should look at controlling calls via the FreeSWITCH event > socket instead of from the dialplan. The nice thing about the event > socket is that your call control can happen on a separate machine. There > is a PHP module for the ESL (event socket library) and it would be > relatively easy for you to get going. Here are some links to get you > started: > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > http://wiki.freeswitch.org/wiki/Event_Socket > > If you absolutely MUST have call control with scripts inside of the > dialplan then there simply is no better choice than Lua. You can learn > Lua in a few hours, but getting mod_php finished and debugged will take > time, money, and other resources that no one seems willing to spend. > Here is some information to consider: > > http://wiki.freeswitch.org/wiki/Mod_lua > > Come join us on IRC (#freeswitch on irc.freenode.net > ) if you want to discuss this further. > > -MC (IRC: mercutioviz) > > 2009/6/13 Christian L?schenkohl > > > hello > > i am working for an austrian voip carrier. > for a few months i work with freeswitch and it is simply great. > it solves our needs in many places (high volume, flexible, stable). > the only thing i really miss is the avalibilty of php as a call control > language. > mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't > that good (or even there :-) ). > i know there is perl, i also implemented some applications (conference > system with provisioning, > inbound call routing to our application servers, some tests as pbx), but > what should i say - > perl and me aren't compatible in the end. > > the greatest thing for us would be that mod_php comes alive again with > the functional state > of mod_perl > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > there is also a bounty entry about mod_php, to pay for this would also > be an option and > could be discussed. > > keep on with the excellent work and greetings from austria > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung& Entwicklung VoIP > > xpirio > Telekommunikation& Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From darklion11 at yahoo.com Mon Jun 15 02:47:15 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 02:47:15 -0700 (PDT) Subject: [Freeswitch-users] Asterisks to Freeswitch CALL REJECTED Message-ID: <24031735.post@talk.nabble.com> I am trying to call Freeswitch using Asterisks and using a softphone X-Lite but the issue is call rejected by freeswitch? Is their any configuration files to allow asterisks to call to freeswitch? -- View this message in context: http://www.nabble.com/Asterisks-to-Freeswitch-CALL-REJECTED-tp24031735p24031735.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 02:49:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 02:49:04 -0700 (PDT) Subject: [Freeswitch-users] Asterisks to Freeswitch CALL REJECTED Message-ID: <24031735.post@talk.nabble.com> I am trying to call Freeswitch using Asterisks and using a softphone X-Lite but the issue is call rejected by freeswitch? Is their any configuration files to allow asterisks to call to freeswitch? Separate OS am using Linux for Asterisks Windows for freeswitch... -- View this message in context: http://www.nabble.com/Asterisks-to-Freeswitch-CALL-REJECTED-tp24031735p24031735.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 02:59:28 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 02:59:28 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? Message-ID: <24031900.post@talk.nabble.com> is there any available gui for freeswitch using cake php complete instead of wikipbx, spice softphone or pfsense? -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Mon Jun 15 03:06:32 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 18:06:32 +0800 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24031900.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> Message-ID: http://www.tcapi.org/index.php?title=Main_Page On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: > > is there any available gui for freeswitch using cake php complete > instead of > wikipbx, spice softphone or pfsense? > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From darklion11 at yahoo.com Mon Jun 15 03:21:45 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 03:21:45 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: References: <24031900.post@talk.nabble.com> Message-ID: <24032171.post@talk.nabble.com> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully developed... Is there any GUI with billing options? seven-8 wrote: > > http://www.tcapi.org/index.php?title=Main_Page > > > > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: > >> >> is there any available gui for freeswitch using cake php complete >> instead of >> wikipbx, spice softphone or pfsense? >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From durk.debeer at isp.solcon.nl Mon Jun 15 03:26:17 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Mon, 15 Jun 2009 12:26:17 +0200 Subject: [Freeswitch-users] funny effect after minimizing xml files Message-ID: Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect /d -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/b9436da7/attachment.html From dujinfang at gmail.com Mon Jun 15 03:31:20 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 15 Jun 2009 18:31:20 +0800 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24032171.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> Message-ID: <7E486E4D-B08F-4BB3-A1D2-FD97DBACF204@gmail.com> On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote: > > Yup tcapi is a great cake php GUI for freeswitch but it is not yet > fully > developed... > Is there any GUI with billing options? > > AFAIK, no fully developed GUI available yet, just curious, why are you finding a GUI instead of wikipbx or pfsense? > seven-8 wrote: >> >> http://www.tcapi.org/index.php?title=Main_Page >> >> >> >> On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >> >>> >>> is there any available gui for freeswitch using cake php complete >>> instead of >>> wikipbx, spice softphone or pfsense? >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Mon Jun 15 03:35:02 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 15 Jun 2009 20:35:02 +1000 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> Message-ID: <20090615103502.GA32625@jdc.jasonjgw.net> Paul Mahler wrote: > I have a large project coming up. I'm interested in using Freeswitch > instead of SER and Asterisk. > > What is the current status of Freeswitch? Can I safely use it in a > large scale commercial environment? How active is the Freeswitch > developer community? Others have already addressed most of your questions. I would like to point out, however, that the FreeSWITCH developers offer support, by way of a consulting service, on a commercial basis. If you're running FreeSWITCH in a commercial setting and encounter complex issues that require expert advice or attention from developers, consider entering into a consulting contract. This will also help to fund the project and ensure that the development community remains as active as we all want it to be. From jason at jasonjgw.net Mon Jun 15 03:38:08 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 15 Jun 2009 20:38:08 +1000 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: <200906151030.n5FATaG5015367@opera.rednote.net> References: <200906151030.n5FATaG5015367@opera.rednote.net> Message-ID: <20090615103808.GB32625@jdc.jasonjgw.net> Durk de Beer wrote: > > Hello I've minimized de xml files where possible to make a dialplan that is > as short as possible. Now do I've this funny effect to dial my extensions > who are running from 200 to 207. It seams that I'm able to dial an > extension in closed in a number. So for instants if I dial 120275 extension > 202 will ring even tried it whit two extensions in a number like 202205 . > This results in the first extension ringing so 202205, 202 will ring > 205202, 205 will ring. At this time I'm unable to pinpoint the cause of > this behaviour. Could someone point me to the cause of this effect I don't understand the problem, but my general advice is this: learn to read the FreeSWITCH logs carefully. Make sure that the log level is set to "debug", as it is in the default configuration, then carefully check the log files to see which dialplan extension matched and how the call was processed. From dujinfang at gmail.com Mon Jun 15 03:46:18 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 15 Jun 2009 18:46:18 +0800 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: <4a36232b.8b13f30a.3877.3cafSMTPIN_ADDED@mx.google.com> References: <4a36232b.8b13f30a.3877.3cafSMTPIN_ADDED@mx.google.com> Message-ID: how to help without seeing your dialplan? On Jun 15, 2009, at 6:26 PM, Durk de Beer wrote: > Hello I've minimized de xml files where possible to make a dialplan > that is as short as possible. Now do I've this funny effect to dial > my extensions who are running from 200 to 207. It seams that I'm > able to dial an extension in closed in a number. So for instants if > I dial 120275 extension 202 will ring even tried it whit two > extensions in a number like 202205 . This results in the first > extension ringing so 202205, 202 will ring 205202, 205 will ring. At > this time I'm unable to pinpoint the cause of this behaviour. Could > someone point me to the cause of this effect > > /d > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/48d87c9c/attachment-0002.html From dave at 3c.co.uk Mon Jun 15 03:53:20 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 15 Jun 2009 13:53:20 +0300 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: <20090615103122.9F69D20169@smtp.3c.co.uk> References: <20090615103122.9F69D20169@smtp.3c.co.uk> Message-ID: <1245063200.4694.7.camel@dk-d820> You've probably deleted the start/end markers from your dialplan matches..? It might be easier to help if you posted or pastebinned your dialplan. --Dave > Hello I've minimized de xml files where possible to make a dialplan > that is as short as possible. Now do I've this funny effect to dial my > extensions who are running from 200 to 207. It seams that I'm able to > dial an extension in closed in a number. So for instants if I dial > 120275 extension 202 will ring even tried it whit two extensions in a > number like 202205 . This results in the first extension ringing so > 202205, 202 will ring 205202, 205 will ring. At this time I'm unable > to pinpoint the cause of this behaviour. Could someone point me to the > cause of this effect > > > /d > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From saeedahmad1981 at gmail.com Mon Jun 15 04:59:50 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Mon, 15 Jun 2009 13:59:50 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> Message-ID: Michael: We are using 5.0, and i think we tested this feature quiet a while ago and there was no CDR problem. Raymond: Thanks for hint i'll try it... On Fri, Jun 12, 2009 at 6:54 PM, Michael Giagnocavo wrote: > Well, Nextone for instance has a database the keeps most of the state of > calls, and it?s replicated between the two nodes. (I seem to recall the > database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, > the CDRs still get truncated when there?s any kind of switchover. > > > > But Nextone is a closed system with limited services. As MikeJ mentioned, > it was discussed for FS, but it?s a LOT of work to get that state > synchronized. And, every custom app/module would have to register and > support recreating their state. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Saeed Ahmed > *Sent:* Friday, June 12, 2009 7:39 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques > > > > No idea at all, > > It?s a commercial SBC. > > I wish if we can have same functionality in FS. > > > > - Saeed > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Even Andr? > Fiskvik > *Sent:* Friday, June 12, 2009 3:04 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques > > > > Can you comment some more on how this is configured? > > Would it be something that could be added to the wiki in the SBC setup > page? > > > > Best regards, > > Even Andr? Fiskvik > > > > On 12. juni. 2009, at 12.16, Saeed Ahmad wrote: > > > > I've experience with a commercial SBC, these are two machines running in > cluster mode. In that case if one SBC is going down then other will take all > new calls including the call which were active on broken SBC (SIP only). > > Thats quite ideal for wholesale traffic where the SBC will never be idle. > > On Fri, Jun 12, 2009 at 8:25 AM, John Dalgliesh wrote: > > On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote: > > On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote: > > >> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote: > >>> > >>> Well, if you're running multiple machines, waiting for it to drainstop > >>> isn't that big of a deal unless you're in some sort of hurry, right? > >>> Give it an hour or so to drainstop, then kill 'em. > >> > >> Yes that's exactly what I'm trying to do. The problem is some people > will > >> only try one IP address. > > > > Clients that don't properly implement SRV/NAPTR and fail over need to be > > smacked. :) (not customers but software that fails to do that) > > Yes I'm sure much of their software can do this but it has been set up for > static numeric IPs. And getting the IP changed is a week-long process for > some customers! > > > >>> Would it not be simpler to try to do something with re-invites or > REFER, > >>> assuming your endpoints support it? > >> > >> That was actually plan A. I already added a property in sip_profile > called > >> failover_redirect, which specifies another server to try if FS can't > >> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), > >> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max > >> Calls In Progress. > > > > You can't send a 302 to a call thats already established. > > Yes and I don't want to touch established calls - those calls can stay > there until they drop. This is sent to new requests when > switch_core_session_request fails in mod_sofia. > > > >> Turns out not all my endpoints support it :( > > > > AKA broken endpoints. :) > > Some are broken. Some just have this feature disabled. For 'security > reasons'. You know the drill. > > > {P^/ > John > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e08ebdf7/attachment-0002.html From gmaruzz at celliax.org Mon Jun 15 05:16:55 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 15 Jun 2009 14:16:55 +0200 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> Message-ID: <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> Hi Jingwel, thanks for reporting. Could you please add a Jira issue with as much details as possible? general guide for reporting bugs: http://wiki.freeswitch.org/wiki/Reporting_Bugs what to add for skypiax: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests mod_skypiax Jira: http://jira.freeswitch.org/browse/MODSKYPIAX Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang wrote: > Hi Team, > > I've been using the record_session feature to record call sessions. Here's > how I prepared the dialplan: > > ??? > ????? > ??????? > ??????? > ????? > ??? > > And here's how I trigger it: > > ??? freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA > 2909/userBBB > > The call can be established and the data.wav file was generated without any > problem. However, once userAAA hung up, a segmentation fault occurred and > freeswitch was automatically shut down. Here are what I got from the > console: > > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA > 2909/userBBB > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > Ring-Ready skypiax/skypiax2/userAAA > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() > Channel [skypiax/skypiax2/userAAA] has been answered > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b > > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] > mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH->2909/userBBB > in context default > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > Ring-Ready skypiax/ANY/userBBB! > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() > Channel [skypiax/ANY/userBBB] has been answered > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA > [CS_DESTROY] > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] > Segmentation fault (core dumped) > > Please kindly let me know whether there's anything wrong with the dialplan > or the way how I originated the call. > > Thanks! > -Jingwei > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From asannucci at gmail.com Mon Jun 15 06:32:56 2009 From: asannucci at gmail.com (bakko) Date: Mon, 15 Jun 2009 15:32:56 +0200 Subject: [Freeswitch-users] Asterisks to Freeswitch CALL REJECTED In-Reply-To: <24031735.post@talk.nabble.com> References: <24031735.post@talk.nabble.com> Message-ID: <5F950C77D31F44AC96320EDE62F052DE@voztovoice> Hi, if you understand spanish look at: http://www.freeswitch.es/node/61 Regards From anthony.minessale at gmail.com Mon Jun 15 06:59:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Jun 2009 08:59:13 -0500 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <4A361708.8020808@xpirio.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> Message-ID: <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> Did you actually use ESL with the php wrapper when you tried? You can do all those things from outbound event socket fairly easily. That mod_php you saw, never worked it was just a stub and it didn't actually ever work when the guy who added it totally disappeared, I removed it from tree. And you can still do event socket over localhost on the same box if you so choose. If you really want a mod_php it's entirely possible but it would probably cost you upwards of 5k in development costs. 2009/6/15 Christian L?schenkohl > hi > > thank you very much for your input > i can say for me that i realy tried hard to use the event socket library, > but untill now i can't use it like i used all the agi scripts or even > mod_perl now. > > what i do most - in examples, if the server get's an incomming call > > - find the right user for the number (not that easy because of did in > austria), > from database or file > - build the right dial string for the bridge application (here i miss all > the php > string functions most) > - unsing mod_php functions like setVariable, getVariable, answer, transfer, > sleep > (i don't see how to do this with the php esl) > - or i check if the number is part of a conferencing product and build the > right > conference setup > > i think this would also be possible with lua and luasql, but i developed > years with > phpagi und i'm very used to php in every kind of scripting or > how-to-get-a-solution > situation (since over 10 years now). > > for me in our setup it's also the highest goal to get the servers mostly > independent > of each other. i think nobody of our costumers should be unreachable > because a central > scripting/event server or also database server has gone away (as developers > this happens > more often as we would like it to :-)) > > do not get me wrong, freeswitch is very powerfull and in the near future it > will replace > nearly all of our asterisk servers. > > in combination with php the freeswitch plattform would be heaven for me > > i also thought Brian Fertig has some source written (as posted on > http://wiki.freeswitch.org/wiki/Mod_php), > in combination of the mod_python rewrite (page was last modified in june > 2007). > > br > > > On 2009-06-14 01:15, Nik Middleton wrote: > > I couldn?t agree more. We?re working with a group that are developing a > > massive PHP based music application. They are experts in PHP and MySQL > > but not in VOIP/Telephony. By tuning an abstraction layer that uses PHP > > to communicate with the FS event socket, allows them to work on the > > areas they know best and not worry about the telephony side too much. > > We went the lua route, and don?t use the dial plan at all. My view is > > to keep all db access and processing out of FS as much as possible. With > > the event socket you simply don?t need to embed anything apart from the > > essentials. > > > > We are now processing 100,000+ call setups a day (4 hours) per server > > all using php scripts to drive the application. We may well ultimately > > use C++ instead of PHP for the event socket comms, but right now PHP > > does just fine. > > > > Regards > > > > ------------------------------------------------------------------------ > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > > *Michael Collins > > *Sent:* 13 June 2009 21:57 > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > Perhaps you should look at controlling calls via the FreeSWITCH event > > socket instead of from the dialplan. The nice thing about the event > > socket is that your call control can happen on a separate machine. There > > is a PHP module for the ESL (event socket library) and it would be > > relatively easy for you to get going. Here are some links to get you > > started: > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > If you absolutely MUST have call control with scripts inside of the > > dialplan then there simply is no better choice than Lua. You can learn > > Lua in a few hours, but getting mod_php finished and debugged will take > > time, money, and other resources that no one seems willing to spend. > > Here is some information to consider: > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > ) if you want to discuss this further. > > > > -MC (IRC: mercutioviz) > > > > 2009/6/13 Christian L?schenkohl > > > > > > hello > > > > i am working for an austrian voip carrier. > > for a few months i work with freeswitch and it is simply great. > > it solves our needs in many places (high volume, flexible, stable). > > the only thing i really miss is the avalibilty of php as a call control > > language. > > mod_php or mod_freehp are not compiling anymore and my c++ knowledge > isn't > > that good (or even there :-) ). > > i know there is perl, i also implemented some applications (conference > > system with provisioning, > > inbound call routing to our application servers, some tests as pbx), but > > what should i say - > > perl and me aren't compatible in the end. > > > > the greatest thing for us would be that mod_php comes alive again with > > the functional state > > of mod_perl > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > there is also a bounty entry about mod_php, to pay for this would also > > be an option and > > could be discussed. > > > > keep on with the excellent work and greetings from austria > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung& Entwicklung VoIP > > > > xpirio > > Telekommunikation& Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/a926c87a/attachment-0002.html From brian at freeswitch.org Mon Jun 15 07:21:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 09:21:02 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> Message-ID: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> To: <"user" Can you reproduce this or let us in your box to look at it... someone else reported this but I have yet to be able to reproduce it. /b On Jun 15, 2009, at 2:41 AM, seven wrote: > Hi, > > I'm on version 13524, call from zoiper is ok, but when call zoiper, > it keep rejecting calls, anyone can help? I'm seems always not the > right time join in IRC :( > > http://pastebin.freeswitch.org/9383 > > > Thanks. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/d8b5d39b/attachment-0002.html From dujinfang at gmail.com Mon Jun 15 07:37:23 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 22:37:23 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: Yes, I can reproduce this, FYI, I have another box runs 13272 with the same zoiper without any problem. You can login to our box, I will find you on IRC. On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e04a36ae/attachment-0002.html From mitul at enterux.com Mon Jun 15 07:38:48 2009 From: mitul at enterux.com (Mitul Limbani) Date: Mon, 15 Jun 2009 20:08:48 +0530 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> Message-ID: Anthm, I actually compiled ESL on PHP but wasnt able to figure out how to use it, too little documentation. Can any one throw more light on ESL? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 15-Jun-09, at 19:29, Anthony Minessale wrote: > Did you actually use ESL with the php wrapper when you tried? > You can do all those things from outbound event socket fairly easily. > > That mod_php you saw, never worked it was just a stub and it didn't > actually ever work > when the guy who added it totally disappeared, I removed it from tree. > > And you can still do event socket over localhost on the same box if > you so choose. > > If you really want a mod_php it's entirely possible but it would > probably cost you upwards > of 5k in development costs. > > > 2009/6/15 Christian L?schenkohl > hi > > thank you very much for your input > i can say for me that i realy tried hard to use the event socket > library, > but untill now i can't use it like i used all the agi scripts or > even mod_perl now. > > what i do most - in examples, if the server get's an incomming call > > - find the right user for the number (not that easy because of did > in austria), > from database or file > - build the right dial string for the bridge application (here i > miss all the php > string functions most) > - unsing mod_php functions like setVariable, getVariable, answer, > transfer, sleep > (i don't see how to do this with the php esl) > - or i check if the number is part of a conferencing product and > build the right > conference setup > > i think this would also be possible with lua and luasql, but i > developed years with > phpagi und i'm very used to php in every kind of scripting or how-to- > get-a-solution > situation (since over 10 years now). > > for me in our setup it's also the highest goal to get the servers > mostly independent > of each other. i think nobody of our costumers should be unreachable > because a central > scripting/event server or also database server has gone away (as > developers this happens > more often as we would like it to :-)) > > do not get me wrong, freeswitch is very powerfull and in the near > future it will replace > nearly all of our asterisk servers. > > in combination with php the freeswitch plattform would be heaven for > me > > i also thought Brian Fertig has some source written (as posted on http://wiki.freeswitch.org/wiki/Mod_php > ), > in combination of the mod_python rewrite (page was last modified in > june 2007). > > br > > > On 2009-06-14 01:15, Nik Middleton wrote: > > I couldn?t agree more. We?re working with a group that are > developing a > > massive PHP based music application. They are experts in PHP and > MySQL > > but not in VOIP/Telephony. By tuning an abstraction layer that > uses PHP > > to communicate with the FS event socket, allows them to work on the > > areas they know best and not worry about the telephony side too > much. > > We went the lua route, and don?t use the dial plan at all. My vie > w is > > to keep all db access and processing out of FS as much as > possible. With > > the event socket you simply don?t need to embed anything apart fro > m the > > essentials. > > > > We are now processing 100,000+ call setups a day (4 hours) per > server > > all using php scripts to drive the application. We may well > ultimately > > use C++ instead of PHP for the event socket comms, but right now PHP > > does just fine. > > > > Regards > > > > > --- > --------------------------------------------------------------------- > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > > *Michael Collins > > *Sent:* 13 June 2009 21:57 > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > Perhaps you should look at controlling calls via the FreeSWITCH > event > > socket instead of from the dialplan. The nice thing about the event > > socket is that your call control can happen on a separate machine. > There > > is a PHP module for the ESL (event socket library) and it would be > > relatively easy for you to get going. Here are some links to get you > > started: > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > If you absolutely MUST have call control with scripts inside of the > > dialplan then there simply is no better choice than Lua. You can > learn > > Lua in a few hours, but getting mod_php finished and debugged will > take > > time, money, and other resources that no one seems willing to spend. > > Here is some information to consider: > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > ) if you want to discuss this further. > > > > -MC (IRC: mercutioviz) > > > > 2009/6/13 Christian L?schenkohl > > > > > > hello > > > > i am working for an austrian voip carrier. > > for a few months i work with freeswitch and it is simply great. > > it solves our needs in many places (high volume, flexible, stable). > > the only thing i really miss is the avalibilty of php as a call > control > > language. > > mod_php or mod_freehp are not compiling anymore and my c++ > knowledge isn't > > that good (or even there :-) ). > > i know there is perl, i also implemented some applications > (conference > > system with provisioning, > > inbound call routing to our application servers, some tests as > pbx), but > > what should i say - > > perl and me aren't compatible in the end. > > > > the greatest thing for us would be that mod_php comes alive again > with > > the functional state > > of mod_perl > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > there is also a bounty entry about mod_php, to pay for this would > also > > be an option and > > could be discussed. > > > > keep on with the excellent work and greetings from austria > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung& Entwicklung VoIP > > > > xpirio > > Telekommunikation& Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > --- > --------------------------------------------------------------------- > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/b0c2f4a7/attachment-0002.html From christian.loeschenkohl at xpirio.com Mon Jun 15 07:55:39 2009 From: christian.loeschenkohl at xpirio.com (=?windows-1252?Q?Christian_L=F6schenkohl?=) Date: Mon, 15 Jun 2009 16:55:39 +0200 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> Message-ID: <4A3660EB.4070203@xpirio.com> i tried to i think i tried everthing and looked closely to everything in libs/esl/php (of course i build it and included the ESL.php file) but i do not get the idea in complete, does i work in a client-server way or in inbound mode like i want to (that is exactly my point) no examples are there (i would put them in the wiki if i had one) some simple code i would expect wot work, but i doesn't execute("setVariable", "codec_string=PCMA"); $esl->execute("answer"); $esl->execute("sleep", "2"); $esl->execute("streamFile", "/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello"); $esl->execute("hangup", "16"); ?> can you please help me, what do i get wrong? br On 2009-06-15 15:59, Anthony Minessale wrote: > Did you actually use ESL with the php wrapper when you tried? > You can do all those things from outbound event socket fairly easily. > > That mod_php you saw, never worked it was just a stub and it didn't > actually ever work > when the guy who added it totally disappeared, I removed it from tree. > > And you can still do event socket over localhost on the same box if you > so choose. > > If you really want a mod_php it's entirely possible but it would > probably cost you upwards > of 5k in development costs. > > > 2009/6/15 Christian L?schenkohl > > > hi > > thank you very much for your input > i can say for me that i realy tried hard to use the event socket > library, > but untill now i can't use it like i used all the agi scripts or > even mod_perl now. > > what i do most - in examples, if the server get's an incomming call > > - find the right user for the number (not that easy because of did > in austria), > from database or file > - build the right dial string for the bridge application (here i > miss all the php > string functions most) > - unsing mod_php functions like setVariable, getVariable, answer, > transfer, sleep > (i don't see how to do this with the php esl) > - or i check if the number is part of a conferencing product and > build the right > conference setup > > i think this would also be possible with lua and luasql, but i > developed years with > phpagi und i'm very used to php in every kind of scripting or > how-to-get-a-solution > situation (since over 10 years now). > > for me in our setup it's also the highest goal to get the servers > mostly independent > of each other. i think nobody of our costumers should be unreachable > because a central > scripting/event server or also database server has gone away (as > developers this happens > more often as we would like it to :-)) > > do not get me wrong, freeswitch is very powerfull and in the near > future it will replace > nearly all of our asterisk servers. > > in combination with php the freeswitch plattform would be heaven for me > > i also thought Brian Fertig has some source written (as posted on > http://wiki.freeswitch.org/wiki/Mod_php), > in combination of the mod_python rewrite (page was last modified in > june 2007). > > br > > > On 2009-06-14 01:15, Nik Middleton wrote: > > I couldn?t agree more. We?re working with a group that are > developing a > > massive PHP based music application. They are experts in PHP and > MySQL > > but not in VOIP/Telephony. By tuning an abstraction layer that > uses PHP > > to communicate with the FS event socket, allows them to work on the > > areas they know best and not worry about the telephony side too much. > > We went the lua route, and don?t use the dial plan at all. My > view is > > to keep all db access and processing out of FS as much as > possible. With > > the event socket you simply don?t need to embed anything apart > from the > > essentials. > > > > We are now processing 100,000+ call setups a day (4 hours) per server > > all using php scripts to drive the application. We may well > ultimately > > use C++ instead of PHP for the event socket comms, but right now PHP > > does just fine. > > > > Regards > > > > > ------------------------------------------------------------------------ > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of > > *Michael Collins > > *Sent:* 13 June 2009 21:57 > > *To:* freeswitch-users at lists.freeswitch.org > > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > Perhaps you should look at controlling calls via the FreeSWITCH event > > socket instead of from the dialplan. The nice thing about the event > > socket is that your call control can happen on a separate > machine. There > > is a PHP module for the ESL (event socket library) and it would be > > relatively easy for you to get going. Here are some links to get you > > started: > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > If you absolutely MUST have call control with scripts inside of the > > dialplan then there simply is no better choice than Lua. You can > learn > > Lua in a few hours, but getting mod_php finished and debugged > will take > > time, money, and other resources that no one seems willing to spend. > > Here is some information to consider: > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > > ) if you want to discuss this further. > > > > -MC (IRC: mercutioviz) > > > > 2009/6/13 Christian L?schenkohl > > > >> > > > > hello > > > > i am working for an austrian voip carrier. > > for a few months i work with freeswitch and it is simply great. > > it solves our needs in many places (high volume, flexible, stable). > > the only thing i really miss is the avalibilty of php as a call > control > > language. > > mod_php or mod_freehp are not compiling anymore and my c++ > knowledge isn't > > that good (or even there :-) ). > > i know there is perl, i also implemented some applications > (conference > > system with provisioning, > > inbound call routing to our application servers, some tests as > pbx), but > > what should i say - > > perl and me aren't compatible in the end. > > > > the greatest thing for us would be that mod_php comes alive again > with > > the functional state > > of mod_perl > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > there is also a bounty entry about mod_php, to pay for this would > also > > be an option and > > could be discussed. > > > > keep on with the excellent work and greetings from austria > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung& Entwicklung VoIP > > > > xpirio > > Telekommunikation& Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From dujinfang at gmail.com Mon Jun 15 08:07:50 2009 From: dujinfang at gmail.com (seven) Date: Mon, 15 Jun 2009 23:07:50 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: Brain, You are not on irc right now and it is midnight so I'm gona to sleep. however when I try to reproduce that I found it event didn't get to Zoiper. I use the same Zoiper login to two boxes at the same time, version 13272 is ok while the other isn't. I noticed there is an extra line on the log of version 13524: version 13272: 2009-06-15 22:54:14 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/ internal/839 SOFIA INIT 2009-06-15 22:54:14 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/ internal/839) State Change CS_INIT -> CS_ROUTING version 13524: 2009-06-15 22:50:46 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/ internal/637 SOFIA INIT 2009-06-15 22:50:46 [DEBUG] sofia_glue.c:1599 sofia_glue_do_invite() sip:637 at 192.168.1.27:5070 %3 Setting proxy route to sofia/internal/637 2009-06-15 22:50:46 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/ internal/637) State Change CS_INIT -> CS_ROUTING And here is a more detailed paste: http://pastebin.freeswitch.org/9386 Thank you taking time for this. If you need more detail I'd like to collect and can open ssh for further debug. 7. On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/4ae227da/attachment-0002.html From anthony.minessale at gmail.com Mon Jun 15 08:07:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Jun 2009 10:07:36 -0500 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <4A3660EB.4070203@xpirio.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> Message-ID: <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> look at the perl examples they should translate to php as all the objects and methods are the same. Does anyone who uses ESL + scripting have any time to toss up some wiki pages? 2009/6/15 Christian L?schenkohl > i tried to > i think i tried everthing and looked closely to everything in libs/esl/php > (of > course i build it and included the ESL.php file) > > but i do not get the idea in complete, does i work in a client-server way > or > in inbound mode like i want to (that is exactly my point) > > no examples are there (i would put them in the wiki if i had one) > some simple code i would expect wot work, but i doesn't > > > require_once("ESL.php"); > $esl = new eslConnection('127.0.0.1', '8021', 'asgag243tsa'); > > $esl->execute("setVariable", "codec_string=PCMA"); > $esl->execute("answer"); > $esl->execute("sleep", "2"); > $esl->execute("streamFile", > "/opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello"); > $esl->execute("hangup", "16"); > > ?> > > can you please help me, what do i get wrong? > > br > > > On 2009-06-15 15:59, Anthony Minessale wrote: > > Did you actually use ESL with the php wrapper when you tried? > > You can do all those things from outbound event socket fairly easily. > > > > That mod_php you saw, never worked it was just a stub and it didn't > > actually ever work > > when the guy who added it totally disappeared, I removed it from tree. > > > > And you can still do event socket over localhost on the same box if you > > so choose. > > > > If you really want a mod_php it's entirely possible but it would > > probably cost you upwards > > of 5k in development costs. > > > > > > 2009/6/15 Christian L?schenkohl > > > > > > hi > > > > thank you very much for your input > > i can say for me that i realy tried hard to use the event socket > > library, > > but untill now i can't use it like i used all the agi scripts or > > even mod_perl now. > > > > what i do most - in examples, if the server get's an incomming call > > > > - find the right user for the number (not that easy because of did > > in austria), > > from database or file > > - build the right dial string for the bridge application (here i > > miss all the php > > string functions most) > > - unsing mod_php functions like setVariable, getVariable, answer, > > transfer, sleep > > (i don't see how to do this with the php esl) > > - or i check if the number is part of a conferencing product and > > build the right > > conference setup > > > > i think this would also be possible with lua and luasql, but i > > developed years with > > phpagi und i'm very used to php in every kind of scripting or > > how-to-get-a-solution > > situation (since over 10 years now). > > > > for me in our setup it's also the highest goal to get the servers > > mostly independent > > of each other. i think nobody of our costumers should be unreachable > > because a central > > scripting/event server or also database server has gone away (as > > developers this happens > > more often as we would like it to :-)) > > > > do not get me wrong, freeswitch is very powerfull and in the near > > future it will replace > > nearly all of our asterisk servers. > > > > in combination with php the freeswitch plattform would be heaven for > me > > > > i also thought Brian Fertig has some source written (as posted on > > http://wiki.freeswitch.org/wiki/Mod_php), > > in combination of the mod_python rewrite (page was last modified in > > june 2007). > > > > br > > > > > > On 2009-06-14 01:15, Nik Middleton wrote: > > > I couldn?t agree more. We?re working with a group that are > > developing a > > > massive PHP based music application. They are experts in PHP and > > MySQL > > > but not in VOIP/Telephony. By tuning an abstraction layer that > > uses PHP > > > to communicate with the FS event socket, allows them to work on > the > > > areas they know best and not worry about the telephony side too > much. > > > We went the lua route, and don?t use the dial plan at all. My > > view is > > > to keep all db access and processing out of FS as much as > > possible. With > > > the event socket you simply don?t need to embed anything apart > > from the > > > essentials. > > > > > > We are now processing 100,000+ call setups a day (4 hours) per > server > > > all using php scripts to drive the application. We may well > > ultimately > > > use C++ instead of PHP for the event socket comms, but right now > PHP > > > does just fine. > > > > > > Regards > > > > > > > > > ------------------------------------------------------------------------ > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > > ] *On Behalf > Of > > > *Michael Collins > > > *Sent:* 13 June 2009 21:57 > > > *To:* freeswitch-users at lists.freeswitch.org > > > > > *Subject:* Re: [Freeswitch-users] mod_php needed > > > > > > Perhaps you should look at controlling calls via the FreeSWITCH > event > > > socket instead of from the dialplan. The nice thing about the > event > > > socket is that your call control can happen on a separate > > machine. There > > > is a PHP module for the ESL (event socket library) and it would be > > > relatively easy for you to get going. Here are some links to get > you > > > started: > > > > > > http://wiki.freeswitch.org/wiki/PHP_Event_Socket > > > http://wiki.freeswitch.org/wiki/Event_Socket > > > > > > If you absolutely MUST have call control with scripts inside of > the > > > dialplan then there simply is no better choice than Lua. You can > > learn > > > Lua in a few hours, but getting mod_php finished and debugged > > will take > > > time, money, and other resources that no one seems willing to > spend. > > > Here is some information to consider: > > > > > > http://wiki.freeswitch.org/wiki/Mod_lua > > > > > > Come join us on IRC (#freeswitch on irc.freenode.net > > > > > ) if you want to discuss this further. > > > > > > -MC (IRC: mercutioviz) > > > > > > 2009/6/13 Christian L?schenkohl > > > > > > > >> > > > > > > hello > > > > > > i am working for an austrian voip carrier. > > > for a few months i work with freeswitch and it is simply great. > > > it solves our needs in many places (high volume, flexible, > stable). > > > the only thing i really miss is the avalibilty of php as a call > > control > > > language. > > > mod_php or mod_freehp are not compiling anymore and my c++ > > knowledge isn't > > > that good (or even there :-) ). > > > i know there is perl, i also implemented some applications > > (conference > > > system with provisioning, > > > inbound call routing to our application servers, some tests as > > pbx), but > > > what should i say - > > > perl and me aren't compatible in the end. > > > > > > the greatest thing for us would be that mod_php comes alive again > > with > > > the functional state > > > of mod_perl > > > (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes). > > > there is also a bounty entry about mod_php, to pay for this would > > also > > > be an option and > > > could be discussed. > > > > > > keep on with the excellent work and greetings from austria > > > > > > -- > > > Ing. Christian L?schenkohl > > > Technische Leitung, Forschung& Entwicklung VoIP > > > > > > xpirio > > > Telekommunikation& Service GmbH > > > Lakeside B04 > > > 9020 Klagenfurt > > > Austria > > > > > > T +43 (0) 5 77 11 - 1000 > > > F +43 (0) 5 77 11 - 1002 > > > E christian.loeschenkohl at xpirio.com > > > > > > > > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > Technische Leitung, Forschung & Entwicklung VoIP > > > > xpirio > > Telekommunikation & Service GmbH > > Lakeside B04 > > 9020 Klagenfurt > > Austria > > > > T +43 (0) 5 77 11 - 1000 > > F +43 (0) 5 77 11 - 1002 > > E christian.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/be55ea82/attachment-0002.html From william.suffill at gmail.com Mon Jun 15 08:12:22 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 15 Jun 2009 11:12:22 -0400 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> Message-ID: <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> Any suggestions of what would be a good example in PHP using ESL to document? I'll take a stab at writing something up this week but it would help to have some idea what would be useful. I've used it and got it working but rather document a generic real life example versus my unique use cases. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/5fea324b/attachment-0002.html From anthony.minessale at gmail.com Mon Jun 15 08:27:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 15 Jun 2009 10:27:33 -0500 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> Message-ID: <191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com> A good start would probably be: example of making an inbound connection from script to FS and execute a FSAPI command, like status or show channels. example of making an inbound connection and listening for events and printing them serialized. example of an outbound socket connection where the call is answered, a variable is set then perhaps play one of the pre-installed files and hangup. That last one could be demonstrated using a native socket server or by using ivrd, a little mini forking daemon I added to listen for socket outbound calls and determine a script from channel variables and call that script assuming to use stdin/stdout as the socket. (kinda like agi's) I think that if everyone pooled their experienced together you could probably produce a wrapper that would allow you to use some of your legacy agi code with ESL, naturally you would have to change the names of the apps and a few other things but there is a lot to build on here. I left this portiion of the system where it is so that the community and how it's most commonly used will drive the direction the top layer of code takes. On Mon, Jun 15, 2009 at 10:12 AM, William Suffill wrote: > Any suggestions of what would be a good example in PHP using ESL to > document? I'll take a stab at writing something up this week but it would > help to have some idea what would be useful. I've used it and got it working > but rather document a generic real life example versus my unique use cases. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1d0b2823/attachment-0002.html From dujinfang at gmail.com Mon Jun 15 08:53:29 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 15 Jun 2009 23:53:29 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: <1322BB82-F5CC-47AD-B686-557CA2147EDC@gmail.com> Hi, the difference is the Contact where the "%3B" should be ";" . Is it configurable or a bug? 13272: Call-ID: OTg4NmRlNzY5OThmNzgwM2E3ZmRkYzVhNjVmODMyYjA. User: 839 at 192.168.1.15 Contact: "user" Agent: Zoiper rev.3065 Status: Registered(UDP)(unknown) EXP(2009-06-16 00:54:12) Host: pbx3.veecue.com IP: 192.168.1.27 Port: 5070 Auth-User: 839 Auth-Realm: 192.168.1.15 13524: Call-ID: NWJhYWY0YjJmMzdlNWQ4MWIwZjc2NGM5NjQzZDU3NTg. User: 637 at 192.168.1.16 Contact: "user" Agent: Zoiper rev.3065 Status: Registered(UDP-NAT)(unknown) EXP(2009-06-16 01:42:56) Host: pbx1.veecue.com IP: 192.168.1.27 Port: 5070 Auth-User: 637 Auth-Realm: 192.168.1.16 On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/94c56399/attachment-0002.html From steve.kurzeja at gmail.com Mon Jun 15 03:17:26 2009 From: steve.kurzeja at gmail.com (Steve Kurzeja) Date: Mon, 15 Jun 2009 22:17:26 +1200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms> <0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org> <45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com> <6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> Message-ID: <5f7152000906150317i451b9f7s5c7a8d4293f7c5d1@mail.gmail.com> On Sat, Jun 13, 2009 at 4:54 AM, Michael Giagnocavo wrote: > Well, Nextone for instance has a database the keeps most of the state of > calls, and it?s replicated between the two nodes. (I seem to recall the > database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, > the CDRs still get truncated when there?s any kind of switchover. > > > BTW in Nextone v4.0.x the GNU db is used for storing configuration data like storing routes & other bits which is then loaded into memory. Nextone 4.3 and above uses postgres for this configuration data. The actual call state information is stored in memory and replicated to the standby box via some custom network protocol. Stateful call migration would be a very useful feature in FS but I imagine its way down the roadmap. But as to the original question of live upgrades, having some form of load balancing proxy and then bleeding off traffic from the box you want to upgrade is the most feasible approach, as others have mentioned. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1cb0e5a7/attachment-0002.html From saeedahmad1981 at gmail.com Mon Jun 15 09:31:28 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 15 Jun 2009 18:31:28 +0200 Subject: [Freeswitch-users] Live Upgrade Techniques In-Reply-To: <5f7152000906150317i451b9f7s5c7a8d4293f7c5d1@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C670262E23ECB@mse17be1.mse17.exchange.ms><0E870CD4-1653-44F7-B301-8754FB18EE2A@freeswitch.org><45C4836A-7E78-46D1-9F86-63C3C7633D29@me.com><6E8D2069C08AA84A83D336E996AE4C670262E24017@mse17be1.mse17.exchange.ms> <5f7152000906150317i451b9f7s5c7a8d4293f7c5d1@mail.gmail.com> Message-ID: <41C915AED64E45869CD719DEEDB8D8BF@saeedlaptop> Yeah I was missing this word: SCM => Stateful Call Migration. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Kurzeja Sent: Monday, June 15, 2009 12:17 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Live Upgrade Techniques On Sat, Jun 13, 2009 at 4:54 AM, Michael Giagnocavo wrote: Well, Nextone for instance has a database the keeps most of the state of calls, and it's replicated between the two nodes. (I seem to recall the database was GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still get truncated when there's any kind of switchover. BTW in Nextone v4.0.x the GNU db is used for storing configuration data like storing routes & other bits which is then loaded into memory. Nextone 4.3 and above uses postgres for this configuration data. The actual call state information is stored in memory and replicated to the standby box via some custom network protocol. Stateful call migration would be a very useful feature in FS but I imagine its way down the roadmap. But as to the original question of live upgrades, having some form of load balancing proxy and then bleeding off traffic from the box you want to upgrade is the most feasible approach, as others have mentioned. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e9b050c1/attachment-0002.html From Richard.Lamkin at mettoni.com Mon Jun 15 11:27:52 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Mon, 15 Jun 2009 19:27:52 +0100 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to an inbound call ? Message-ID: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin Richard.lamkin at mettonigroup.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/5fa3920d/attachment-0002.html From christian.loeschenkohl at xpirio.com Mon Jun 15 12:08:39 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Mon, 15 Jun 2009 21:08:39 +0200 Subject: [Freeswitch-users] mod_php needed In-Reply-To: <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> References: <4A33E8A0.1070708@xpirio.com> <87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> <4A361708.8020808@xpirio.com> <191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> <4A3660EB.4070203@xpirio.com> <191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> <6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> Message-ID: <4A369C37.1040107@xpirio.com> hi could you provide me a simple example? - connect with esl - get uuid - set a variable (e.g. codec_string=PCMA) - answer the channel - playback a file the script ist called from ivrd, if i get it right in the dialplan it's with ivrd started as ./ivrd -h 127.0.0.1 -p 9999 -------------------- in my setup $esl->api("help") works and also $esl->sendRecv("api help") but $esl->execute() does nothing i use version 1.0.4pre8 if it is helpfull br On 2009-06-15 17:12, William Suffill wrote: > Any suggestions of what would be a good example in PHP using ESL to > document? I'll take a stab at writing something up this week but it > would help to have some idea what would be useful. I've used it and got > it working but rather document a generic real life example versus my > unique use cases. > > -- W > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From edpimentl at gmail.com Mon Jun 15 12:25:28 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 15 Jun 2009 15:25:28 -0400 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" Message-ID: <9dc4a1670906151225x39126633g4ba159d03dab47d0@mail.gmail.com> FYI Now you can directly connect to MySql via javascript http://jsext.webloji.net/ http://jsext.sourceforge.net/JSEXT1.Mysql.html http://www.brainonfire.net/blog/ssjs-on-ubuntu/ And if you wish you had a V8, check out Google SSJS Engine BTW: JS is now 7x faster than it last year. -E http://Gpro.ws http://twitter.com/edpimentl http://facebook.com/facevalu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/22a83214/attachment-0002.html From ron.freeswitch at mcleodnet.com Mon Jun 15 12:50:14 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Mon, 15 Jun 2009 12:50:14 -0700 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> Message-ID: Something to consider is how long will be PSTN allow the call to remain un-answered. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Lamkin Sent: Monday, June 15, 2009 11:28 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin Richard.lamkin at mettonigroup.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/2524086e/attachment-0002.html From brian at freeswitch.org Mon Jun 15 13:24:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 15:24:02 -0500 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> Message-ID: Survey says ... "execute the ring_ready application" /b On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote: > Something to consider is how long will be PSTN allow the call to > remain un-answered. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Richard Lamkin > Sent: Monday, June 15, 2009 11:28 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to > aninbound call ? > > I have a setup where I have a variety of SIP inbound calls > (originated from PSTN) coming from a SIP provider. The SIP lines > are single lines registered with the provider. The provider is > running with a Nortel CS2K. > > I am putting together a simple event driven operator attendant > console and I would like to set up a call queuing system where the > incoming calls are not answered until an operator is ready to > accept a call. I want the operator to know that a call is in the > ringing Q and who it is from. I do not want to auto answer the call > and put them in a MOH Q because the originator will be charged as > soon as the call is answered. > > My question is how do I get a SIP 180 ringing to be sent to an > inbound call and put that call in a Q? The CS2k does convert > ringing on inbound calls to media towards the originator. I?ve > looked through the wiki for examples but not found what I need in > either in dial plan or fifo operations. > > Any help would be gratefully appreciated. > > Regards > > Richard Lamkin > Richard.lamkin at mettonigroup.com > > ************************************************************************* > Please consider the environment before printing this e-mail > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/cdf91f10/attachment-0002.html From lon at kickasspixels.com Mon Jun 15 14:46:01 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 15 Jun 2009 14:46:01 -0700 Subject: [Freeswitch-users] Gateway conference handshake trouble Message-ID: <5d3e0dc60906151446h3258ca4cld28fe1ca9daef8bd@mail.gmail.com> Hi, We are trying to work freeswitch into an older system as a conference bridge. Our existing gateways can hand off the call and they pass some DTMF signals to route everything. Currently, the gateway sends * and then freeswitch returns a # to accept the call for the conference. When that transaction is done the gateway waits 2 seconds to send either a 0 or 1 and indicate if the caller is a moderator/admin of the conference. Everything works until we get to the moderator flag. That part never appears in the logs or debug info in the console. But the person on the call, can hit 0 or 1 and then enter the conference call correctly. If they don't it will timeout and drop them. >From what we know the gateway is sending the moderator DTMF flag, but freeswitch is only listening on the call for it. Any idea would appreciated. Lon From stevecrozz at gmail.com Mon Jun 15 14:51:30 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 15 Jun 2009 14:51:30 -0700 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <9dc4a1670906151225x39126633g4ba159d03dab47d0@mail.gmail.com> References: <9dc4a1670906151225x39126633g4ba159d03dab47d0@mail.gmail.com> Message-ID: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> What a great project, does anyone know what's needed to make these libraries available to freeswitch scripts? --Stephen On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl wrote: > FYI > Now you can directly connect to MySql via javascript > > http://jsext.webloji.net/ > http://jsext.sourceforge.net/JSEXT1.Mysql.html > http://www.brainonfire.net/blog/ssjs-on-ubuntu/ > > And if you wish you had a V8, check out Google SSJS Engine > BTW: JS is now 7x faster than it last year. > > -E > http://Gpro.ws > http://twitter.com/edpimentl > http://facebook.com/facevalu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/2e15f5cd/attachment-0002.html From krice at freeswitch.org Mon Jun 15 14:58:46 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 15 Jun 2009 16:58:46 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> Message-ID: You know you can already access any sql support by UnixODBC via mod_spidermonkey already... And NO its not slow (maybe that was true 10 years ago, but not any longer) From: Stephen Crosby Reply-To: Date: Mon, 15 Jun 2009 14:51:30 -0700 To: Subject: Re: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" What a great project, does anyone know what's needed to make these libraries available to freeswitch scripts? --Stephen On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl wrote: > FYI > Now you can directly connect to MySql via javascript > > http://jsext.webloji.net/ > http://jsext.sourceforge.net/JSEXT1.Mysql.html > http://www.brainonfire.net/blog/ssjs-on-ubuntu/ > > And if you wish you had a V8, check out Google SSJS Engine > BTW: JS is now 7x faster than it last year. > > -E > http://Gpro.ws > http://twitter.com/edpimentl > http://facebook.com/facevalu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/d7532d6b/attachment-0002.html From stevecrozz at gmail.com Mon Jun 15 15:08:12 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 15 Jun 2009 15:08:12 -0700 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> Message-ID: <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> I'm actually much more interested in the HTTP library and a few other components than MySQL. Freeswitch's spidermonkey CURL library doesn't provide returned HTTP status codes and JSEXT does. That said, I'm still somewhat interested in the mysql library over odbc. For me, the only thing I've ever really done with ODBC is use it to bridge the gap between freeswitch mod_spidermonkey and mysql server. I think it would be nice to not need it. --Stephen On Mon, Jun 15, 2009 at 2:58 PM, Ken Rice wrote: > You know you can already access any sql support by UnixODBC via > mod_spidermonkey already... And NO its not slow (maybe that was true 10 > years ago, but not any longer) > > > ------------------------------ > *From: *Stephen Crosby > *Reply-To: * > *Date: *Mon, 15 Jun 2009 14:51:30 -0700 > *To: * > *Subject: *Re: [Freeswitch-users] Access MySQL directly via Javascript > using SSJS Engines .... "really!" > > > What a great project, does anyone know what's needed to make these > libraries available to freeswitch scripts? > > --Stephen > > On Mon, Jun 15, 2009 at 12:25 PM, EdPimentl wrote: > > FYI > Now you can directly connect to MySql via javascript > > http://jsext.webloji.net/ > http://jsext.sourceforge.net/JSEXT1.Mysql.html > http://www.brainonfire.net/blog/ssjs-on-ubuntu/ > > And if you wish you had a V8, check out Google SSJS Engine > BTW: JS is now 7x faster than it last year. > > -E > http://Gpro.ws > http://twitter.com/edpimentl > http://facebook.com/facevalu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/e041de66/attachment-0002.html From brian at freeswitch.org Mon Jun 15 15:23:04 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 17:23:04 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> Message-ID: <538DFEC7-F2BB-45C0-837D-A5DB0B9F889D@freeswitch.org> On Jun 15, 2009, at 5:08 PM, Stephen Crosby wrote: > I'm actually much more interested in the HTTP library and a few > other components than MySQL. Freeswitch's spidermonkey CURL library > doesn't provide returned HTTP status codes and JSEXT does.\\\ Patch it! ;) > > > That said, I'm still somewhat interested in the mysql library over > odbc. For me, the only thing I've ever really done with ODBC is use > it to bridge the gap between freeswitch mod_spidermonkey and mysql > server. I think it would be nice to not need it. > > --Stephen From edpimentl at gmail.com Mon Jun 15 15:45:47 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 15 Jun 2009 18:45:47 -0400 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> Message-ID: <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> Actually these new SSJS engines such GoogleV8 and other such as JAXER Bring a entire new way of building robust webapp/desktop app/ mobile app like it has never been built before... For those that love Google GWT=Java_To_Javascript and dislike verbosity of Java, there is PyJamas ... Google GWT=Python_To_Javascript JS today is not the same it was years ago or even months ago. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/ce481122/attachment-0002.html From brian at freeswitch.org Mon Jun 15 16:35:00 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 18:35:00 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> Message-ID: <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> I don't think V8 will work on 64bit yet will it? /b On Jun 15, 2009, at 5:45 PM, EdPimentl wrote: > Actually these new SSJS engines such GoogleV8 and other such as JAXER > Bring a entire new way of building robust webapp/desktop app/ mobile > app like it has never been built before... > > For those that love Google GWT=Java_To_Javascript and dislike > verbosity of Java, > there is PyJamas ... Google GWT=Python_To_Javascript > > JS today is not the same it was years ago or even months ago. > > -E From Prometheus001 at gmx.net Mon Jun 15 17:19:57 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 02:19:57 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? Message-ID: <4A36E52D.2010005@gmx.net> I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter From brian at freeswitch.org Mon Jun 15 17:27:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 15 Jun 2009 19:27:44 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A36E52D.2010005@gmx.net> References: <4A36E52D.2010005@gmx.net> Message-ID: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: > > What is the best way to have this done? Move the call to park and then > retransfer again with intercom, or is there a better solution? From darklion11 at yahoo.com Mon Jun 15 18:31:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 18:31:13 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <7E486E4D-B08F-4BB3-A1D2-FD97DBACF204@gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <7E486E4D-B08F-4BB3-A1D2-FD97DBACF204@gmail.com> Message-ID: <24045733.post@talk.nabble.com> I like to use this GUI for both OS windows and linux. Wikipbx and PFSENSE is for linux only... Just a simple website that I need to integrate... seven-8 wrote: > > > On Jun 15, 2009, at 6:21 PM, Edmar Cruz wrote: > >> >> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >> fully >> developed... >> Is there any GUI with billing options? >> >> > > AFAIK, no fully developed GUI available yet, just curious, why are you > finding a GUI instead of wikipbx or pfsense? > > >> seven-8 wrote: >>> >>> http://www.tcapi.org/index.php?title=Main_Page >>> >>> >>> >>> On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>> >>>> >>>> is there any available gui for freeswitch using cake php complete >>>> instead of >>>> wikipbx, spice softphone or pfsense? >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24045733.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 18:38:18 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 18:38:18 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24045824.post@talk.nabble.com> I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 15 18:43:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 18:43:41 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Message-ID: <24045890.post@talk.nabble.com> Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER = root PASS = password ..... I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24045890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Jun 15 19:28:23 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 15 Jun 2009 21:28:23 -0500 Subject: [Freeswitch-users] Is Freeswitch ready for prime time? In-Reply-To: <1245058387.4694.5.camel@dk-d820> References: <80952EFB-6327-44B2-8A7C-FC1467E61666@ringcarrier.com> <1245058387.4694.5.camel@dk-d820> Message-ID: <13DE8568-1F4D-4224-B07B-77C03B844726@freeswitch.org> Hehe, where can I buy stock in this company? :) -MC Sent from my iPhone On Jun 15, 2009, at 4:33 AM, David Knell wrote: > On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote: >> What is the current status of Freeswitch? Can I safely use it in a >> large scale commercial environment? How active is the Freeswitch >> developer community? > > Hi Paul - > > We've used FS over the last 18 months or so to handle millions of > calls > - some wholesale in/out, some IVR, some calling card, some > callthrough - > with a total value in the millions of dollars; we have no complaints. > > --Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From diego.viola at gmail.com Mon Jun 15 19:56:31 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 15 Jun 2009 22:56:31 -0400 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24032171.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> Message-ID: <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> I'm currently writing a rails app that uses mod_nibblebill for billing, it's a calling card app. Diego On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz wrote: > > Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully > developed... > Is there any GUI with billing options? > > > seven-8 wrote: > > > > http://www.tcapi.org/index.php?title=Main_Page > > > > > > > > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: > > > >> > >> is there any available gui for freeswitch using cake php complete > >> instead of > >> wikipbx, spice softphone or pfsense? > >> -- > >> View this message in context: > >> > http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/d243308c/attachment-0002.html From edpimentl at gmail.com Mon Jun 15 19:58:09 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 15 Jun 2009 22:58:09 -0400 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> Message-ID: <9dc4a1670906151958o4914b94cr46ac4ba313ba7315@mail.gmail.com> As of April 09 it did not support 64bit .... not sure if it has been added since then. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090615/1325a8ba/attachment-0002.html From darklion11 at yahoo.com Mon Jun 15 20:06:00 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 20:06:00 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> Message-ID: <24046873.post@talk.nabble.com> Can you share me the link of it so i can try... Please Diego Viola wrote: > > I'm currently writing a rails app that uses mod_nibblebill for billing, > it's > a calling card app. > > Diego > > On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz wrote: > >> >> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully >> developed... >> Is there any GUI with billing options? >> >> >> seven-8 wrote: >> > >> > http://www.tcapi.org/index.php?title=Main_Page >> > >> > >> > >> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >> > >> >> >> >> is there any available gui for freeswitch using cake php complete >> >> instead of >> >> wikipbx, spice softphone or pfsense? >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 15 22:14:40 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 00:14:40 -0500 Subject: [Freeswitch-users] Access MySQL directly via Javascript using SSJS Engines .... "really!" In-Reply-To: <9dc4a1670906151958o4914b94cr46ac4ba313ba7315@mail.gmail.com> References: <11990ade0906151451l6a24e275ue7d84985414cc46@mail.gmail.com> <11990ade0906151508o5c1f8ab8qccc1e82e3556108f@mail.gmail.com> <9dc4a1670906151545y359d426bl45ac90824adf97a2@mail.gmail.com> <54BBF083-1082-424D-9467-60C41E8205B9@freeswitch.org> <9dc4a1670906151958o4914b94cr46ac4ba313ba7315@mail.gmail.com> Message-ID: <30E63C81-EAAF-407B-9B50-E05BD5EB1B1C@freeswitch.org> Pretty useless without 64bit support. /b On Jun 15, 2009, at 9:58 PM, EdPimentl wrote: > As of April 09 it did not support 64bit .... not sure if it has been > added since then. > -E From darklion11 at yahoo.com Mon Jun 15 23:08:39 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 15 Jun 2009 23:08:39 -0700 (PDT) Subject: [Freeswitch-users] created external5090 on profile not working? Message-ID: <24048269.post@talk.nabble.com> I created a profile name external5090 on /usr/local/freeswitch/conf/sip_profiles/external5090.xml... Change ext-sip-ip and ext-rtp-ip for a server 192.168.0.104 with sip-port: 5090... My local Ip is 192.168.0.105... I see it with I type it on the API freeswitch and type sofia status is there... How can I know that it is working? can u send me a API freeswitch for it? may code is originate sofia/external5090/1002 at 192.168.0.104:5090 5090 is this correct? -- View this message in context: http://www.nabble.com/created-external5090-on-profile-not-working--tp24048269p24048269.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From durk.debeer at isp.solcon.nl Mon Jun 15 23:38:22 2009 From: durk.debeer at isp.solcon.nl (Durk de Beer) Date: Tue, 16 Jun 2009 08:38:22 +0200 Subject: [Freeswitch-users] funny effect after minimizing xml files In-Reply-To: References: Message-ID: >> >> Hello I've minimized de xml files where possible to make a dialplan that is >> as short as possible. Now do I've this funny effect to dial my extensions >> who are running from 200 to 207. It seams that I'm able to dial an >> extension in closed in a number. So for instants if I dial 120275 extension >> 202 will ring even tried it whit two extensions in a number like 202205 . >> This results in the first extension ringing so 202205, 202 will ring >> 205202, 205 will ring. At this time I'm unable to pinpoint the cause of >> this behaviour. Could someone point me to the cause of this effect > I don't understand the problem, but my general advice is this: learn to read > the FreeSWITCH logs carefully. Make sure that the log level is set to "debug", > as it is in the default configuration, then carefully check the log files to > see which dialplan extension matched and how the call was processed. After reading this, a colleague of mine had a look at the logs and found out that we had goofed up the regular expressions in the dialplan. This made Freeswitch dial the number of an extension if its sequence was found in the dialed number so lets say the extension has number 202 and the number dialed was 15320264 it would find the 202 sequence in the dialed number and then it would dial the 202 extension. So it seems this one is a stupid mistake of us. Thanks to all that responded \d From darklion11 at yahoo.com Tue Jun 16 00:44:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 00:44:31 -0700 (PDT) Subject: [Freeswitch-users] Is there anyone who is connected to PCCW? Message-ID: <24049302.post@talk.nabble.com> PCCW is use for making calls through IP connected through cellphone just enter the areacode for example 900639274522123 900-prefix 63-areacode 9274522123 - number? Has anyone has tried it? Please help me how to connect to it -- View this message in context: http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Tue Jun 16 01:15:51 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 16 Jun 2009 16:15:51 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> Message-ID: <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> Hi Giovanni, I've reported it in Jira. Here's the bug url: http://jira.freeswitch.org/browse/MODSKYPIAX-35 Thanks, -Jingwei On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli wrote: > Hi Jingwel, > thanks for reporting. > > Could you please add a Jira issue with as much details as possible? > > general guide for reporting bugs: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > what to add for skypiax: > > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests > > mod_skypiax Jira: > http://jira.freeswitch.org/browse/MODSKYPIAX > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang > wrote: > > Hi Team, > > > > I've been using the record_session feature to record call sessions. > Here's > > how I prepared the dialplan: > > > > > > > > > > > > > > > > > > And here's how I trigger it: > > > > freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA > > 2909/userBBB > > > > The call can be established and the data.wav file was generated without > any > > problem. However, once userAAA hung up, a segmentation fault occurred and > > freeswitch was automatically shut down. Here are what I got from the > > console: > > > > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA > > 2909/userBBB > > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 > switch_channel_set_name() > > New Channel skypiax/skypiax2/userAAA > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] > > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > > Ring-Ready skypiax/skypiax2/userAAA > > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 > outbound_channel_answered() > > Channel [skypiax/skypiax2/userAAA] has been answered > > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 > switch_ivr_session_transfer() > > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] > > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: > > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b > > > > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] > > mod_dialplan_xml.c:252 dialplan_hunt() Processing > FreeSWITCH->2909/userBBB > > in context default > > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 > switch_channel_set_name() > > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] > > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() > > Ring-Ready skypiax/ANY/userBBB! > > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 > outbound_channel_answered() > > Channel [skypiax/ANY/userBBB] has been answered > > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 > > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA > [CS_EXECUTE] > > [NORMAL_CLEARING] > > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 > > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA > > [CS_DESTROY] > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended > > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > > switch_core_session_thread() Close Channel skypiax/ANY/userBBB > [CS_DESTROY] > > Segmentation fault (core dumped) > > > > Please kindly let me know whether there's anything wrong with the > dialplan > > or the way how I originated the call. > > > > Thanks! > > -Jingwei > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/bb8fe2e9/attachment-0002.html From gmaruzz at celliax.org Tue Jun 16 01:42:53 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 16 Jun 2009 10:42:53 +0200 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> Message-ID: <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> Hi Jingwei, Thanks a lot! I'll take care of as soon as possible. Btw, before I read the Jira, are you testing in linux? If you are testing on linux, would you please report how it is performing under load? I mean, what is the load average with, let say, 10 or 20 or more concurrent Skype call? This has nothing to do with your bug, but will help me in getting better performances. Ciao for now, and thanks again for reporting! -giovanni On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang wrote: > Hi Giovanni, > > I've reported it in Jira. Here's the bug url: > > http://jira.freeswitch.org/browse/MODSKYPIAX-35 > > Thanks, > -Jingwei > > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli > wrote: >> >> Hi Jingwel, >> thanks for reporting. >> >> Could you please add a Jira issue with as much details as possible? >> >> general guide for reporting bugs: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> what to add for skypiax: >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >> >> mod_skypiax Jira: >> http://jira.freeswitch.org/browse/MODSKYPIAX >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang >> wrote: >> > Hi Team, >> > >> > I've been using the record_session feature to record call sessions. >> > Here's >> > how I prepared the dialplan: >> > >> > ??? >> > ????? >> > ??????? >> > ??????? >> > ????? >> > ??? >> > >> > And here's how I trigger it: >> > >> > ??? freeswitch at localhost.localdomain>originate skypiax/skypiax2/userAAA >> > 2909/userBBB >> > >> > The call can be established and the data.wav file was generated without >> > any >> > problem. However, once userAAA hung up, a segmentation fault occurred >> > and >> > freeswitch was automatically shut down. Here are what I got from the >> > console: >> > >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA >> > 2909/userBBB >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >> > switch_channel_set_name() >> > New Channel skypiax/skypiax2/userAAA >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >> > remote_party_is_ringing() >> > Ring-Ready skypiax/skypiax2/userAAA >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >> > outbound_channel_answered() >> > Channel [skypiax/skypiax2/userAAA] has been answered >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >> > switch_ivr_session_transfer() >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >> > >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >> > FreeSWITCH->2909/userBBB >> > in context default >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >> > switch_channel_set_name() >> > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >> > remote_party_is_ringing() >> > Ring-Ready skypiax/ANY/userBBB! >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >> > outbound_channel_answered() >> > Channel [skypiax/ANY/userBBB] has been answered >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >> > [CS_EXECUTE] >> > [NORMAL_CLEARING] >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >> > [CS_DESTROY] >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >> > [CS_DESTROY] >> > Segmentation fault (core dumped) >> > >> > Please kindly let me know whether there's anything wrong with the >> > dialplan >> > or the way how I originated the call. >> > >> > Thanks! >> > -Jingwei >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Tue Jun 16 02:17:00 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 05:17:00 -0400 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24046873.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> Message-ID: <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> I'm currently rewriting the entire thing, it was a commercial app first, but I'm re-writing it in order to make it open source. It's not ready yet, as soon as I finish it, I will release it to the public. Diego On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: > > Can you share me the link of it so i can try... Please > > Diego Viola wrote: >> >> I'm currently writing a rails app that uses mod_nibblebill for billing, >> it's >> a calling card app. >> >> Diego >> >> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz wrote: >> >>> >>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully >>> developed... >>> Is there any GUI with billing options? >>> >>> >>> seven-8 wrote: >>> > >>> > http://www.tcapi.org/index.php?title=Main_Page >>> > >>> > >>> > >>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>> > >>> >> >>> >> is there any available gui for freeswitch using cake php complete >>> >> instead of >>> >> wikipbx, spice softphone or pfsense? >>> >> -- >>> >> View this message in context: >>> >> >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >> >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Tue Jun 16 02:24:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 11:24:11 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: References: <4A36E52D.2010005@gmx.net> Message-ID: <4A3764BB.3040403@gmx.net> Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: > click on the AA button? :) > > /b > > On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: > > >> What is the best way to have this done? Move the call to park and then >> retransfer again with intercom, or is there a better solution? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Tue Jun 16 02:26:29 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 02:26:29 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> Message-ID: <24050713.post@talk.nabble.com> Thanks for that info... Can you send me this project if and only if it is already finished on this email darklion at yahoo.com? Thanks a lot... Diego Viola wrote: > > I'm currently rewriting the entire thing, it was a commercial app > first, but I'm re-writing it in order to make it open source. It's not > ready yet, as soon as I finish it, I will release it to the public. > > Diego > > On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: >> >> Can you share me the link of it so i can try... Please >> >> Diego Viola wrote: >>> >>> I'm currently writing a rails app that uses mod_nibblebill for billing, >>> it's >>> a calling card app. >>> >>> Diego >>> >>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>> wrote: >>> >>>> >>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>> fully >>>> developed... >>>> Is there any GUI with billing options? >>>> >>>> >>>> seven-8 wrote: >>>> > >>>> > http://www.tcapi.org/index.php?title=Main_Page >>>> > >>>> > >>>> > >>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>> > >>>> >> >>>> >> is there any available gui for freeswitch using cake php complete >>>> >> instead of >>>> >> wikipbx, spice softphone or pfsense? >>>> >> -- >>>> >> View this message in context: >>>> >> >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Tue Jun 16 02:32:21 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 16 Jun 2009 17:32:21 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> Message-ID: <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> Sure, I'll append to you the result tomorrow. Regards, -Jingwei On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli wrote: > Hi Jingwei, > > Thanks a lot! I'll take care of as soon as possible. > > Btw, before I read the Jira, are you testing in linux? > > If you are testing on linux, would you please report how it is > performing under load? I mean, what is the load average with, let say, > 10 or 20 or more concurrent Skype call? > > This has nothing to do with your bug, but will help me in getting > better performances. > > Ciao for now, and thanks again for reporting! > > -giovanni > > > > > On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang > wrote: > > Hi Giovanni, > > > > I've reported it in Jira. Here's the bug url: > > > > http://jira.freeswitch.org/browse/MODSKYPIAX-35 > > > > Thanks, > > -Jingwei > > > > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Hi Jingwel, > >> thanks for reporting. > >> > >> Could you please add a Jira issue with as much details as possible? > >> > >> general guide for reporting bugs: > >> http://wiki.freeswitch.org/wiki/Reporting_Bugs > >> > >> what to add for skypiax: > >> > >> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests > >> > >> mod_skypiax Jira: > >> http://jira.freeswitch.org/browse/MODSKYPIAX > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> ========================================= > >> www.celliax.org > >> via Pierlombardo 9, 20135 Milano > >> Italy > >> gmaruzz at celliax dot org > >> Cell : +39-347-2665618 > >> Fax : +39-02-87390039 > >> > >> > >> > >> > >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang > >> wrote: > >> > Hi Team, > >> > > >> > I've been using the record_session feature to record call sessions. > >> > Here's > >> > how I prepared the dialplan: > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > And here's how I trigger it: > >> > > >> > freeswitch at localhost.localdomain>originate > skypiax/skypiax2/userAAA > >> > 2909/userBBB > >> > > >> > The call can be established and the data.wav file was generated > without > >> > any > >> > problem. However, once userAAA hung up, a segmentation fault occurred > >> > and > >> > freeswitch was automatically shut down. Here are what I got from the > >> > console: > >> > > >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA > >> > 2909/userBBB > >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 > >> > switch_channel_set_name() > >> > New Channel skypiax/skypiax2/userAAA > >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] > >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 > >> > remote_party_is_ringing() > >> > Ring-Ready skypiax/skypiax2/userAAA > >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 > >> > outbound_channel_answered() > >> > Channel [skypiax/skypiax2/userAAA] has been answered > >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 > >> > switch_ivr_session_transfer() > >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] > >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: > >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b > >> > > >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] > >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing > >> > FreeSWITCH->2909/userBBB > >> > in context default > >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 > >> > switch_channel_set_name() > >> > New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] > >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 > >> > remote_party_is_ringing() > >> > Ring-Ready skypiax/ANY/userBBB! > >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 > >> > outbound_channel_answered() > >> > Channel [skypiax/ANY/userBBB] has been answered > >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 > >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA > >> > [CS_EXECUTE] > >> > [NORMAL_CLEARING] > >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 > >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB > >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) > Ended > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA > >> > [CS_DESTROY] > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 > >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended > >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 > >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB > >> > [CS_DESTROY] > >> > Segmentation fault (core dumped) > >> > > >> > Please kindly let me know whether there's anything wrong with the > >> > dialplan > >> > or the way how I originated the call. > >> > > >> > Thanks! > >> > -Jingwei > >> > > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/d090ae23/attachment-0002.html From diego.viola at gmail.com Tue Jun 16 03:01:12 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 06:01:12 -0400 Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24050713.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> <24050713.post@talk.nabble.com> Message-ID: <86a32abc0906160301p802a71arb8d67bc9a0ebcfa9@mail.gmail.com> Sure, I will let you know when it's done. On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruz wrote: > > Thanks for that info... Can you send me this project if and only if it is > already finished on this email darklion at yahoo.com? Thanks a lot... > > > Diego Viola wrote: >> >> I'm currently rewriting the entire thing, it was a commercial app >> first, but I'm re-writing it in order to make it open source. It's not >> ready yet, as soon as I finish it, I will release it to the public. >> >> Diego >> >> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: >>> >>> Can you share me the link of it so i can try... Please >>> >>> Diego Viola wrote: >>>> >>>> I'm currently writing a rails app that uses mod_nibblebill for billing, >>>> it's >>>> a calling card app. >>>> >>>> Diego >>>> >>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>>> fully >>>>> developed... >>>>> Is there any GUI with billing options? >>>>> >>>>> >>>>> seven-8 wrote: >>>>> > >>>>> > http://www.tcapi.org/index.php?title=Main_Page >>>>> > >>>>> > >>>>> > >>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>>> > >>>>> >> >>>>> >> is there any available gui for freeswitch using cake php complete >>>>> >> instead of >>>>> >> wikipbx, spice softphone or pfsense? >>>>> >> -- >>>>> >> View this message in context: >>>>> >> >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Tue Jun 16 03:20:06 2009 From: dujinfang at gmail.com (seven) Date: Tue, 16 Jun 2009 18:20:06 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: Hi brain, Are you still looking into this? I think it must be some error when it register, I manually changed the contract str in the registration db, immediately it works. After re- register, stop work again. Should I report this to jira? sqlite> select contact from sip_registrations where contact like '%637%'; contact "user" sqlite> update sip_registrations set contact='"user" ' where contact like '%637%'; On Jun 15, 2009, at 10:21 PM, Brian West wrote: > To: <"user" > > Can you reproduce this or let us in your box to look at it... > someone else reported this but I have yet to be able to reproduce it. > > /b > > On Jun 15, 2009, at 2:41 AM, seven wrote: > >> Hi, >> >> I'm on version 13524, call from zoiper is ok, but when call zoiper, >> it keep rejecting calls, anyone can help? I'm seems always not the >> right time join in IRC :( >> >> http://pastebin.freeswitch.org/9383 >> >> >> Thanks. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/80a991b7/attachment-0002.html From darklion11 at yahoo.com Tue Jun 16 03:59:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 03:59:01 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <24050713.post@talk.nabble.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> <24050713.post@talk.nabble.com> Message-ID: <24051970.post@talk.nabble.com> Hello sir, Do you know how to connect to two freeswitch at a time with different Ip addresses? If a user is register on FreeSwitch 1, the user should not have another account or he/she will not register anymore for Freeswitch 2? They can call each other... I already make one but an error occur Can't find user 1566331 at 192.168.0.105 You must define a domain called "192.168.0.105" in your directory and add a user="1566331" ..... Can you give me an example? Thanks for the help. Edmar Cruz wrote: > > Thanks for that info... Can you send me this project if and only if it is > already finished on this email darklion at yahoo.com? Thanks a lot... > > > Diego Viola wrote: >> >> I'm currently rewriting the entire thing, it was a commercial app >> first, but I'm re-writing it in order to make it open source. It's not >> ready yet, as soon as I finish it, I will release it to the public. >> >> Diego >> >> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote: >>> >>> Can you share me the link of it so i can try... Please >>> >>> Diego Viola wrote: >>>> >>>> I'm currently writing a rails app that uses mod_nibblebill for billing, >>>> it's >>>> a calling card app. >>>> >>>> Diego >>>> >>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>>> fully >>>>> developed... >>>>> Is there any GUI with billing options? >>>>> >>>>> >>>>> seven-8 wrote: >>>>> > >>>>> > http://www.tcapi.org/index.php?title=Main_Page >>>>> > >>>>> > >>>>> > >>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>>> > >>>>> >> >>>>> >> is there any available gui for freeswitch using cake php complete >>>>> >> instead of >>>>> >> wikipbx, spice softphone or pfsense? >>>>> >> -- >>>>> >> View this message in context: >>>>> >> >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24051970.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue Jun 16 04:35:02 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 06:35:02 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> Message-ID: <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> If you can catch brian or me on irc can you provide remote access to this box and we should be able to fix this pretty quick Mike On Jun 16, 2009, at 5:20 AM, seven wrote: > Hi brain, > > Are you still looking into this? > > I think it must be some error when it register, I manually changed > the contract str in the registration db, immediately it works. > After re-register, stop work again. > > Should I report this to jira? > > sqlite> select contact from sip_registrations where contact like > '%637%'; > contact > "user" 637@ > 192.168.1.27: > 5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip > %3A637%40192.168.1.27%3A5070%3Brinstance > %3D8df223525ea557b0%3Btransport%3DUDP> > > sqlite> update sip_registrations set contact='"user" :5070;rinstance=8df223525ea557b0;transport=UDP;fs_nat=yes;fs_path=sip > %3A637%40192.168.1.27%3A5070>' where contact like '%637%'; > > > > > On Jun 15, 2009, at 10:21 PM, Brian West wrote: >> To: <"user" >> >> Can you reproduce this or let us in your box to look at it... >> someone else reported this but I have yet to be able to reproduce it. >> >> /b >> >> On Jun 15, 2009, at 2:41 AM, seven wrote: >> >>> Hi, >>> >>> I'm on version 13524, call from zoiper is ok, but when call >>> zoiper, it keep rejecting calls, anyone can help? I'm seems always >>> not the right time join in IRC :( >>> >>> http://pastebin.freeswitch.org/9383 >>> >>> >>> Thanks. >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/50718e0b/attachment-0002.html From mike at jerris.com Tue Jun 16 04:50:39 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 06:50:39 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A3764BB.3040403@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> Message-ID: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX wrote: > Hello Brian, > > this is too easy :-). > > This is for a small callcenter app and I only want the user to pickup > the call once (to accept the call in X-Lite (or a Snom phone) and to > start the workflow on the web application). I do not want him to > accept > the call on the phone and then on the Web app. > > Best regards > Peter > > > > Brian West schrieb: >> click on the AA button? :) >> >> /b >> >> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >> >> >>> What is the best way to have this done? Move the call to park and >>> then >>> retransfer again with intercom, or is there a better solution? >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Tue Jun 16 04:55:02 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 16 Jun 2009 07:55:02 -0400 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A3764BB.3040403@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> Message-ID: <4A378816.8020504@freeswitch.org> Peter P GMX wrote: > Hello Brian, > > this is too easy :-). > > This is for a small callcenter app and I only want the user to pickup > the call once (to accept the call in X-Lite (or a Snom phone) and to > start the workflow on the web application). I do not want him to accept > the call on the phone and then on the Web app. > is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray From gcd at i.ph Tue Jun 16 05:02:58 2009 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 16 Jun 2009 20:02:58 +0800 Subject: [Freeswitch-users] Is there anyone who is connected to PCCW? In-Reply-To: <24049302.post@talk.nabble.com> References: <24049302.post@talk.nabble.com> Message-ID: <7d0bfd8c0906160502u7d3f6decxa1dc667a44eb6c4@mail.gmail.com> what is PCCW? could you please fill in more details what you like to do. to connect mobile phones w/ FS, the mobile phone has to have SIP feature. pls search the Wiki for some models. -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Tue, Jun 16, 2009 at 3:44 PM, Edmar Cruz wrote: > > PCCW is use for making calls through IP connected through cellphone just > enter the areacode for example > > 900639274522123 > > 900-prefix > 63-areacode > 9274522123 - number? > > Has anyone has tried it? > > Please help me how to connect to it > -- > View this message in context: > http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p24049302.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/9c8171e0/attachment-0002.html From paul.degt at gmail.com Tue Jun 16 05:18:24 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Tue, 16 Jun 2009 08:18:24 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A35C72A.6030804@gmail.com> References: <4A35C72A.6030804@gmail.com> Message-ID: <4A378D90.2040209@gmail.com> Solved by replacing "auto-nat" with public ip in public profile "external_sip-ip" and "extrenal-rtp-ip" params. I believe values for these params used to be taken from vars.xml and so would have public ips by default - would be nice to document such changes in README. paul.degt at gmail.com wrote: > Trying out latest trunk ans seems like js function session.getDigits() > stopped working (not collecting any digits), I do see > > switch_rtp.c:1560 Send end packet for [5] ts=2222260 > dur=2080/2080/2000 seq=8732 > > in debug log so I assume dtmf is ok. > Anybody can shed some light on why wouldn't it work now? > Works just fine under 1.0.3 release. I use slightly modified version > of disa.js from fs examples. > > Thanks. > From Prometheus001 at gmx.net Tue Jun 16 05:32:29 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 14:32:29 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A378816.8020504@freeswitch.org> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A378816.8020504@freeswitch.org> Message-ID: <4A3790DD.8090009@gmx.net> Hello Ray, I do use event socket and it pushes me a link on the website whenever a call for this agent comes in. It's just a matter of visibility. The agent may still finish his old workflow and is still entering data. When a call comes in then and he picks up the phone, the data he just entered is gone away. So I would like the web app to drive answering the call. It gives a better visibility about what he is doing to the callcenter agent. Best regards Peter Raymond Chandler schrieb: > Peter P GMX wrote: > >> Hello Brian, >> >> this is too easy :-). >> >> This is for a small callcenter app and I only want the user to pickup >> the call once (to accept the call in X-Lite (or a Snom phone) and to >> start the workflow on the web application). I do not want him to accept >> the call on the phone and then on the Web app. >> >> > is there any reason you don't make your web app listen to event socket > or event sink to catch the answer event and start the workflow? then you > just need to answer the call on the softphone and the webapp should > automatically start the workflow. > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Tue Jun 16 05:38:21 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 14:38:21 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> Message-ID: <4A37923D.8060809@gmx.net> Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: > The only way I can think to do this today would be to cancel the call > and re send with the intercom headers for a phone that supports it. > It may be possible to send a reinvite with autoanswer headers but I > doubt that would work, all you could do is try making code to do it it > a sipp or sipsak scenario and test it. A better aproach might be to > answer the call normally and detect that to start your web workflow or > not really ring the phone, just the web app and deliver the call with > autoanswer when the button is hit in the web ui. > > Mike > > On Jun 16, 2009, at 4:24 AM, Peter P GMX wrote: > > >> Hello Brian, >> >> this is too easy :-). >> >> This is for a small callcenter app and I only want the user to pickup >> the call once (to accept the call in X-Lite (or a Snom phone) and to >> start the workflow on the web application). I do not want him to >> accept >> the call on the phone and then on the Web app. >> >> Best regards >> Peter >> >> >> >> Brian West schrieb: >> >>> click on the AA button? :) >>> >>> /b >>> >>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>> >>> >>> >>>> What is the best way to have this done? Move the call to park and >>>> then >>>> retransfer again with intercom, or is there a better solution? >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dujinfang at gmail.com Tue Jun 16 05:40:00 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 16 Jun 2009 20:40:00 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> Message-ID: <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 637 at 192.168.1.16 Contact: "user" Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. On Jun 16, 2009, at 7:35 PM, Michael Jerris wrote: > If you can catch brian or me on irc can you provide remote access to > this box and we should be able to fix this pretty quick > > Mike > > On Jun 16, 2009, at 5:20 AM, seven wrote: > >> Hi brain, >> >> Are you still looking into this? >> >> I think it must be some error when it register, I manually changed >> the contract str in the registration db, immediately it works. >> After re-register, stop work again. >> >> Should I report this to jira? >> >> sqlite> select contact from sip_registrations where contact like >> '%637%'; >> contact >> "user" > > >> >> sqlite> update sip_registrations set contact='"user" > >' where contact like '%637%'; >> >> >> >> >> On Jun 15, 2009, at 10:21 PM, Brian West wrote: >>> To: <"user" >>> >>> Can you reproduce this or let us in your box to look at it... >>> someone else reported this but I have yet to be able to reproduce >>> it. >>> >>> /b >>> >>> On Jun 15, 2009, at 2:41 AM, seven wrote: >>> >>>> Hi, >>>> >>>> I'm on version 13524, call from zoiper is ok, but when call >>>> zoiper, it keep rejecting calls, anyone can help? I'm seems >>>> always not the right time join in IRC :( >>>> >>>> http://pastebin.freeswitch.org/9383 >>>> >>>> >>>> Thanks. >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/715283ea/attachment-0002.html From brian at freeswitch.org Tue Jun 16 05:41:04 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 07:41:04 -0500 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A378D90.2040209@gmail.com> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> Message-ID: Can you please put it back to auto-nat and email me the output of global_getvar from the CLI so I can see what it detected? /b On Jun 16, 2009, at 7:18 AM, paul.degt at gmail.com wrote: > Solved by replacing "auto-nat" with public ip in public profile > "external_sip-ip" and "extrenal-rtp-ip" params. > I believe values for these params used to be taken from vars.xml and > so > would have public ips by default - would be nice to document such > changes in README. From brian at freeswitch.org Tue Jun 16 05:43:24 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 07:43:24 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37923D.8060809@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> Message-ID: Why not just keep the agent off hook.. in park state... then just playback ringing before you bridge? /b On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: > Hello Michael, > > I want the phone be ringing, just for acoustical feedback reasons. > > But what if I > > * transfer it to the same user destination again (now with intercom > enabled), will this work? > * transfer it to park and then transfer it to the same destination > again (now with intercom enabled) > > Best regards > Peter From brian at freeswitch.org Tue Jun 16 05:43:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 07:43:53 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> Message-ID: <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: > Almost caught you on IRC Mike. > > Our server is in a NAT'd network and all agents registered in the > same LAN. I can remotely register by using the public IP and the > contact string is right. > > Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. > User: 637 at 192.168.1.16 > Contact: "user" > > Agent: Zoiper rev.1809 > > So it's like only happens on our LAN and where there's a fs_path > present. > > Just curious, why agents registered on a local LAN has param > fs_nat=yes; (default internal profile, port 5060) ? > > Seems our time doesn't match, I'm generally available in office > 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/84dc0224/attachment-0002.html From asannucci at gmail.com Tue Jun 16 05:51:12 2009 From: asannucci at gmail.com (bakko) Date: Tue, 16 Jun 2009 14:51:12 +0200 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: <24045890.post@talk.nabble.com> References: <24045890.post@talk.nabble.com> Message-ID: Did you compiled freeswitch with this command? ./configure --enable-core-odbc-support makemake installRegards From dujinfang at gmail.com Tue Jun 16 06:03:40 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 16 Jun 2009 21:03:40 +0800 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: References: <24045890.post@talk.nabble.com> Message-ID: <575E18F3-9021-4482-8CB6-C6799C6EDB7E@gmail.com> current configure will automatically use odbc if it's available, no need the --enable-core-odbc-support anymore. better to check if unixodbc-dev package installed of not. On Jun 16, 2009, at 8:51 PM, bakko wrote: > Did you compiled freeswitch with this command? > > ./configure --enable-core-odbc-support > makemake installRegards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Tue Jun 16 06:07:21 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 15:07:21 +0200 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> Message-ID: <4A379909.9050400@gmx.net> May this help also: I just tried current Zoiper with TLS. Outbound is working, inbound not. Zoiper registeres with the following contact info: "7233213" When a call comes in, Zoiper rings once and then hangs up. It shows "service or option not implemented" in the Zoiper log. My snom phones with the same parameters in the same network (they are all nated) register differently "723323" My FS logs show for an incoming call to Zoiper: 7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) Running State Change CS_CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) State CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [calling][0] 2009-06-16 14:50:16.340881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [terminated][415] 2009-06-16 14:50:16.340881 [NOTICE] sofia.c:3660 Hangup sofia/internal/sip:7233213 at 217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Its seems that something with the codecs fails here, although I have enabled all codecs in Zoiper and FS offers alaw. Best regards Peter Brian West schrieb: > Ok i'll have to se what I can do about reproducing this issue now that > I have more info on how its happening. > > /b > > On Jun 16, 2009, at 7:40 AM, dujinfang wrote: > >> Almost caught you on IRC Mike. >> >> Our server is in a NAT'd network and all agents registered in the >> same LAN. I can remotely register by using the public IP and the >> contact string is right. >> >> Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. >> User: 637 at 192.168.1.16 >> Contact: "user" >> >> Agent: Zoiper rev.1809 >> >> So it's like only happens on our LAN and where there's a fs_path present. >> >> Just curious, why agents registered on a local LAN has param >> fs_nat=yes; (default internal profile, port 5060) ? >> >> Seems our time doesn't match, I'm generally available in office >> 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. >> >> Thank you. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Tue Jun 16 06:17:25 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 16 Jun 2009 21:17:25 +0800 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> Message-ID: <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> What's wrong of the contact string? 639(snom) works but 637(zoiper) doesn't. "user" "seven" On Jun 16, 2009, at 8:43 PM, Brian West wrote: > Ok i'll have to se what I can do about reproducing this issue now > that I have more info on how its happening. > > /b > > On Jun 16, 2009, at 7:40 AM, dujinfang wrote: > >> Almost caught you on IRC Mike. >> >> Our server is in a NAT'd network and all agents registered in the >> same LAN. I can remotely register by using the public IP and the >> contact string is right. >> >> Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. >> User: 637 at 192.168.1.16 >> Contact: "user" > > >> Agent: Zoiper rev.1809 >> >> So it's like only happens on our LAN and where there's a fs_path >> present. >> >> Just curious, why agents registered on a local LAN has param >> fs_nat=yes; (default internal profile, port 5060) ? >> >> Seems our time doesn't match, I'm generally available in office >> 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. >> >> Thank you. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/cb7ae392/attachment-0002.html From mike at jerris.com Tue Jun 16 06:21:42 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 08:21:42 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37923D.8060809@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> Message-ID: <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: > Hello Michael, > > I want the phone be ringing, just for acoustical feedback reasons. > > But what if I > > * transfer it to the same user destination again (now with intercom > enabled), will this work? > * transfer it to park and then transfer it to the same destination > again (now with intercom enabled) > > Best regards > Peter > > Michael Jerris schrieb: >> The only way I can think to do this today would be to cancel the call >> and re send with the intercom headers for a phone that supports it. >> It may be possible to send a reinvite with autoanswer headers but I >> doubt that would work, all you could do is try making code to do it >> it >> a sipp or sipsak scenario and test it. A better aproach might be to >> answer the call normally and detect that to start your web workflow >> or >> not really ring the phone, just the web app and deliver the call with >> autoanswer when the button is hit in the web ui. >> >> Mike >> >> On Jun 16, 2009, at 4:24 AM, Peter P GMX >> wrote: >> >> >>> Hello Brian, >>> >>> this is too easy :-). >>> >>> This is for a small callcenter app and I only want the user to >>> pickup >>> the call once (to accept the call in X-Lite (or a Snom phone) and to >>> start the workflow on the web application). I do not want him to >>> accept >>> the call on the phone and then on the Web app. >>> >>> Best regards >>> Peter >>> >>> >>> >>> Brian West schrieb: >>> >>>> click on the AA button? :) >>>> >>>> /b >>>> >>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>> >>>> >>>> >>>>> What is the best way to have this done? Move the call to park and >>>>> then >>>>> retransfer again with intercom, or is there a better solution? >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Jun 16 06:25:37 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 08:25:37 -0500 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> Message-ID: <9EB1D140-2A4E-4CA5-BE6E-458C1C436087@freeswitch.org> I need sip traces... also can you guys register to my dev box? dev.bkw.org with default user/pass try 1009 thru 1015 please. /b On Jun 16, 2009, at 8:17 AM, Seven Du wrote: > What's wrong of the contact string? 639(snom) works but 637(zoiper) > doesn't. > > "user" > > > "seven" > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/6850d80f/attachment-0002.html From marketing at cluecon.com Tue Jun 16 07:24:20 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 16 Jun 2009 07:24:20 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Volunteers needed! Message-ID: <87f2f3b90906160724m159a3c36t2c840e9305ad59d7@mail.gmail.com> Spread the word! We have need of some volunteers to assist us with various tasks at ClueCon this year. As you may know, when putting on a conference there are numerous little things that require attention. Having several designated volunteers to handle these tasks will make the conference better for everyone. If you or someone you know would like to help out then please email me off list. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/3ca59299/attachment-0002.html From panselva at gmail.com Tue Jun 16 08:17:59 2009 From: panselva at gmail.com (selva kumar) Date: Tue, 16 Jun 2009 20:47:59 +0530 Subject: [Freeswitch-users] Receiving calls us FS (Inbound) Message-ID: <45f609f90906160817i344c74c8i586687d65586f740@mail.gmail.com> Hi, I've tried configuring the inbound settings in default.xml, internal.xml, public.xml and acl.conf.xml. I am trying to route the call to one of the extension let's say 1005. It works well now. However, the outgoing is not happening but it worked find before Inbound is done. Now, when I remove the settings whatever I made to achieve inbound routing, the outbound works well. I am wondering like what needs to be made to achieve to blended environment. i.e. I need to be able to make outbound call and receive incoming calls. Request you to assist me in resolving the problem. Thanks Sam. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/92a5ce67/attachment-0002.html From mike at jerris.com Tue Jun 16 09:14:58 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 12:14:58 -0400 Subject: [Freeswitch-users] Zoiper reject freeswitch calls In-Reply-To: <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> References: <3D324905-47E3-47E0-8940-5E00CBF3C738@gmail.com> <342F31CE-0564-434F-89E6-1F5C70C6E496@freeswitch.org> <4B7333D0-F0CE-4A1B-BADA-B9FDA0602132@jerris.com> <5D434405-4E82-489D-BFB3-33980118291E@gmail.com> <5355A789-DC34-4D63-8CDF-EC30BDA4E795@freeswitch.org> <6DD6BA21-AE4E-4B72-A10C-82D7B2B32160@gmail.com> Message-ID: This issue is now fixed in svn. Thanks Seven for access to your box to troubleshoot. Mike On Jun 16, 2009, at 9:17 AM, Seven Du wrote: > What's wrong of the contact string? 639(snom) works but 637(zoiper) > doesn't. > > "user" > > > "seven" > > > > On Jun 16, 2009, at 8:43 PM, Brian West wrote: >> Ok i'll have to se what I can do about reproducing this issue now >> that I have more info on how its happening. >> >> /b >> >> On Jun 16, 2009, at 7:40 AM, dujinfang wrote: >> >>> Almost caught you on IRC Mike. >>> >>> Our server is in a NAT'd network and all agents registered in the >>> same LAN. I can remotely register by using the public IP and the >>> contact string is right. >>> >>> Call-ID: ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. >>> User: 637 at 192.168.1.16 >>> Contact: "user" >> > >>> Agent: Zoiper rev.1809 >>> >>> So it's like only happens on our LAN and where there's a fs_path >>> present. >>> >>> Just curious, why agents registered on a local LAN has param >>> fs_nat=yes; (default internal profile, port 5060) ? >>> >>> Seems our time doesn't match, I'm generally available in office >>> 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. >>> >>> Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/7aca339a/attachment-0002.html From raul at etellicom.com Tue Jun 16 09:16:05 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 16 Jun 2009 13:16:05 -0300 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A36E52D.2010005@gmx.net> References: <4A36E52D.2010005@gmx.net> Message-ID: <1245168965.11865.35.camel@raul-laptop> I actually do that with our call center application. For all incoming calls, our IVR engine parks the call in a virtual extension and plays back prompts, advertisements, MOH, process digits, etc. When the queue management finds an available agent, it sends an event to the client application for that agent (with an optional screen-pop) where the agent can click "Answer Call" and then we transfer the call with the auto-answer header set on to the agent phone. You could take a similar approach, if you're worrying about only providing ring-back tone to the caller you can simply park the call and use the playback app to play a tone_stream until the agent clicks the web link, which will transfer the call from the parking extension to the agent with the auto-answer flag. I'm still willing to make some tests with REINVITE providing auto-answer headers, as suggested by Mike. That would provide a more generic way to answer calls programmatically when it's already ringing the endpoint. I just need to find some time to read the sofia code and figure out how to do that :) Regards, Raul On Tue, 2009-06-16 at 02:19 +0200, Peter P GMX wrote: > I have managed to have a realtme status of a phone on a web page with > event_socket and a push service to the web bowser. > > What I am now trying to do is roughly the following: > > * when a call comes in, a flashing banner appears on the web page > with an underlying link (this works so far) > * when the user klicks on this flashing banner, the external SIP UA > which is already ringing, shall pick up the call. > > I know that it's possible to autoanswer a call with the intercom > feature. Also the SIP client X-Lite which we use here is able to > autoanswer a call. > I however want to manually decide when the UA takes the call with the > following workflow: > > * X-Lite rings on incoming call > * user klicks on the flashing banner > * X-Lite takes the call > > What is the best way to have this done? Move the call to park and then > retransfer again with intercom, or is there a better solution? > > Best regards > Peter > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Richard.Lamkin at mettoni.com Tue Jun 16 09:41:04 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 16 Jun 2009 17:41:04 +0100 Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to an in bound call ? In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138052FDBCD@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138052FDFE3@nickel.mettonigroup.com> Brian, Thank you for putting me on the right track. I thought I would share my results so after a bit of trial and error testing I came up with the follow DP rule, which lives in dialplan/public/. When an incoming call arrives for DDI 012345678 it is ack'ed with a "180 Ringing" and then the call is held up while the rule goes to sleep. On sleep expiry the call is cleared (from Ron McLeod's comment). This means any incoming call that is not processed using an API method will be automatically cleared after 3 mins. This makes a nice neat way of holing incoming calls ringing. Best Regards Richard Lamkin Richard.lamkin at mettonigroup.com From: Brian West [mailto:brian at freeswitch.org] Sent: 15 June 2009 21:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How do I get a 180 ringing to be sent toaninbound call ? Survey says ... "execute the ring_ready application" /b On Jun 15, 2009, at 2:50 PM, Ron McLeod wrote: Something to consider is how long will be PSTN allow the call to remain un-answered. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Lamkin Sent: Monday, June 15, 2009 11:28 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How do I get a 180 ringing to be sent to aninbound call ? I have a setup where I have a variety of SIP inbound calls (originated from PSTN) coming from a SIP provider. The SIP lines are single lines registered with the provider. The provider is running with a Nortel CS2K. I am putting together a simple event driven operator attendant console and I would like to set up a call queuing system where the incoming calls are not answered until an operator is ready to accept a call. I want the operator to know that a call is in the ringing Q and who it is from. I do not want to auto answer the call and put them in a MOH Q because the originator will be charged as soon as the call is answered. My question is how do I get a SIP 180 ringing to be sent to an inbound call and put that call in a Q? The CS2k does convert ringing on inbound calls to media towards the originator. I've looked through the wiki for examples but not found what I need in either in dial plan or fifo operations. Any help would be gratefully appreciated. Regards Richard Lamkin Richard.lamkin at mettonigroup.com ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/d0571c9e/attachment-0002.html From d at d-man.org Tue Jun 16 09:54:08 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 16 Jun 2009 09:54:08 -0700 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: <24045890.post@talk.nabble.com> References: <24045890.post@talk.nabble.com> Message-ID: What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the real logs from FS's logs? The info below is not nearly detailed enough. -----Original Message----- From: Edmar Cruz [mailto:darklion11 at yahoo.com] Sent: Monday, June 15, 2009 6:44 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER = root PASS = password ..... I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 890p24045890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From d at d-man.org Tue Jun 16 09:59:28 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 16 Jun 2009 09:59:28 -0700 Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed In-Reply-To: <200906091126.03556.yivzhenko@mksat.net> References: <200906091126.03556.yivzhenko@mksat.net> Message-ID: That should not be the case - I will double check this. My apologies if I broke it. :-( Please file a bug on this so I don't forget. _____ From: Yuriy Ivzhenko [mailto:yivzhenko at mksat.net] Sent: Tuesday, June 09, 2009 1:26 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed Some time ago mod_nibblebill was set variable nibble_total_billed after hangup. But after last few updates of module this variable is no more sets. Somebody else have this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/0bb621ef/attachment-0002.html From d at d-man.org Tue Jun 16 09:59:46 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 16 Jun 2009 09:59:46 -0700 Subject: [Freeswitch-users] mod_niible install problem In-Reply-To: References: Message-ID: This should be fixed in the latest build (thanks MikeJ) _____ From: ram [mailto:talk2ram at gmail.com] Sent: Tuesday, June 09, 2009 12:03 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_niible install problem Hi i have downloaded latest SVN and trying to make install i get the following error I googled for the same but there no information on this error how can i resolve this problem Ram making install mod_nibblebill Compiling mod_nibblebill.c... Compiling mod_nibblebill.c ... mod_nibblebill.c: In function ?get_balance?: mod_nibblebill.c:368: error: ?balance? undeclared (first use in this function) mod_nibblebill.c:368: error: (Each undeclared identifier is reported only once mod_nibblebill.c:368: error: for each function it appears in.) make[5]: *** [mod_nibblebill.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_nibblebill-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/1e6c3979/attachment-0002.html From msc at freeswitch.org Tue Jun 16 10:24:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Jun 2009 10:24:11 -0700 Subject: [Freeswitch-users] Receiving calls us FS (Inbound) In-Reply-To: <45f609f90906160817i344c74c8i586687d65586f740@mail.gmail.com> References: <45f609f90906160817i344c74c8i586687d65586f740@mail.gmail.com> Message-ID: <87f2f3b90906161024j382f004et1e33c5611ac7fb4e@mail.gmail.com> I think this question might need to be backed up with some more information. I recommend you post your relevant configs to pastebin so that we can have a look. (pastebin.freeswitch.org) -MC On Tue, Jun 16, 2009 at 8:17 AM, selva kumar wrote: > Hi, > > I've tried configuring the inbound settings in default.xml, internal.xml, > public.xml and acl.conf.xml. > > I am trying to route the call to one of the extension let's say 1005. It > works well now. However, the outgoing is not happening but it worked find > before Inbound is done. > > Now, when I remove the settings whatever I made to achieve inbound routing, > the outbound works well. I am wondering like what needs to be made to > achieve to blended environment. i.e. I need to be able to make outbound call > and receive incoming calls. > > Request you to assist me in resolving the problem. > > Thanks > Sam. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/18ad6ef7/attachment-0002.html From marketing at cluecon.com Tue Jun 16 10:37:23 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 16 Jun 2009 10:37:23 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Getting Ready! Message-ID: <87f2f3b90906161037s44218ecbie96ca23d2df4664b@mail.gmail.com> ClueCon 2009 is only seven weeks away! We are all looking forward to meeting together in Chicago. To make sure that everything goes as planned we would like to know how many people will be attending. If you have not already signed up for ClueCon 2009 please do so. Call 877.742.CLUE and Brian will get you registered. Also, sign up at www.cluecon.com so that you can get updates on speakers, schedules, and sponsors. If you have any questions at all please feel free to call or email us. We look forward to seeing you this August! -Michael Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/b3529a09/attachment-0002.html From Prometheus001 at gmx.net Tue Jun 16 10:49:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 19:49:16 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> Message-ID: <4A37DB1C.8070706@gmx.net> It mainly works now by uuid_transfer the following way via event socket. uuid_setvar sip_invite_params intercom=true uuid_setvar sip_auto_answer true uuid_transfer 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: ;transport=tls;line=er6kxnib Max-Forwards: 68 From: "Peter FS" ;tag=9eQ8rjQy533HF To: Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: Call-Info: ;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: "Peter FS" ;tag=9eQ8rjQy533HF To: ;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: ;reg-id=1 WWW-Authenticate: Digest realm="sip2.mycompany.de", nonce="2ee26efe6ab27f88", algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: > The transfer should work but it sounds like offhook agents is what > your really trying to accomplish so I would go with brian's suggestion. > > > > On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote: > > >> Hello Michael, >> >> I want the phone be ringing, just for acoustical feedback reasons. >> >> But what if I >> >> * transfer it to the same user destination again (now with intercom >> enabled), will this work? >> * transfer it to park and then transfer it to the same destination >> again (now with intercom enabled) >> >> Best regards >> Peter >> >> Michael Jerris schrieb: >> >>> The only way I can think to do this today would be to cancel the call >>> and re send with the intercom headers for a phone that supports it. >>> It may be possible to send a reinvite with autoanswer headers but I >>> doubt that would work, all you could do is try making code to do it >>> it >>> a sipp or sipsak scenario and test it. A better aproach might be to >>> answer the call normally and detect that to start your web workflow >>> or >>> not really ring the phone, just the web app and deliver the call with >>> autoanswer when the button is hit in the web ui. >>> >>> Mike >>> >>> On Jun 16, 2009, at 4:24 AM, Peter P GMX >>> wrote: >>> >>> >>> >>>> Hello Brian, >>>> >>>> this is too easy :-). >>>> >>>> This is for a small callcenter app and I only want the user to >>>> pickup >>>> the call once (to accept the call in X-Lite (or a Snom phone) and to >>>> start the workflow on the web application). I do not want him to >>>> accept >>>> the call on the phone and then on the Web app. >>>> >>>> Best regards >>>> Peter >>>> >>>> >>>> >>>> Brian West schrieb: >>>> >>>> >>>>> click on the AA button? :) >>>>> >>>>> /b >>>>> >>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> What is the best way to have this done? Move the call to park and >>>>>> then >>>>>> retransfer again with intercom, or is there a better solution? >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Tue Jun 16 11:17:01 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 16 Jun 2009 14:17:01 -0400 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37DB1C.8070706@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> <4A37DB1C.8070706@gmx.net> Message-ID: <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> uuid_setvar sip_invite_params intercom=true should be unnecessary. Mike On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: > It mainly works now by uuid_transfer the following way via event > socket. > uuid_setvar sip_invite_params intercom=true > uuid_setvar sip_auto_answer true > uuid_transfer 1000 XML default > so the call is transferred from 1000 to 1000. > > What happens: > 1) If I disable intercom on the Snom phone, the phone rings, stops > ringing and rings again (ok) > 1) If I enable intercom on the Snom phone, the phone rings, stops > ringing and hangs up (not ok) > > So I do not get the Snom to pick up the call in intercom mode. > > The last invite is: > INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib > SIP/2.0 > Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > Route: ;transport=tls;line=er6kxnib > Max-Forwards: 68 > From: "Peter FS" ;tag=9eQ8rjQy533HF > To: > > > Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > CSeq: 116467629 INVITE > Contact: > Call-Info: ;answer-after=0 > The intercom part is there and the Call-Info line with answer-after > also. > > The phone answers with > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > From: "Peter FS" ;tag=9eQ8rjQy533HF > To: > >;tag=71rskygkr2 > Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > CSeq: 116467629 INVITE > Contact: > ;reg-id=1 > WWW-Authenticate: Digest realm="sip2.mycompany.de", > nonce="2ee26efe6ab27f88", algorithm=MD5 > Content-Length: 0 > and hangs up. > > Anybody know how to solve this Snom intercom issue? > > Best regards > Peter > > > Michael Jerris schrieb: >> The transfer should work but it sounds like offhook agents is what >> your really trying to accomplish so I would go with brian's >> suggestion. >> >> >> >> On Jun 16, 2009, at 7:38 AM, Peter P GMX >> wrote: >> >> >>> Hello Michael, >>> >>> I want the phone be ringing, just for acoustical feedback reasons. >>> >>> But what if I >>> >>> * transfer it to the same user destination again (now with >>> intercom >>> enabled), will this work? >>> * transfer it to park and then transfer it to the same destination >>> again (now with intercom enabled) >>> >>> Best regards >>> Peter >>> >>> Michael Jerris schrieb: >>> >>>> The only way I can think to do this today would be to cancel the >>>> call >>>> and re send with the intercom headers for a phone that supports it. >>>> It may be possible to send a reinvite with autoanswer headers but I >>>> doubt that would work, all you could do is try making code to do it >>>> it >>>> a sipp or sipsak scenario and test it. A better aproach might be >>>> to >>>> answer the call normally and detect that to start your web workflow >>>> or >>>> not really ring the phone, just the web app and deliver the call >>>> with >>>> autoanswer when the button is hit in the web ui. >>>> >>>> Mike >>>> >>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX >>>> wrote: >>>> >>>> >>>> >>>>> Hello Brian, >>>>> >>>>> this is too easy :-). >>>>> >>>>> This is for a small callcenter app and I only want the user to >>>>> pickup >>>>> the call once (to accept the call in X-Lite (or a Snom phone) >>>>> and to >>>>> start the workflow on the web application). I do not want him to >>>>> accept >>>>> the call on the phone and then on the Web app. >>>>> >>>>> Best regards >>>>> Peter >>>>> >>>>> >>>>> >>>>> Brian West schrieb: >>>>> >>>>> >>>>>> click on the AA button? :) >>>>>> >>>>>> /b >>>>>> >>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> What is the best way to have this done? Move the call to park >>>>>>> and >>>>>>> then >>>>>>> retransfer again with intercom, or is there a better solution? >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bjbrashier at gmail.com Tue Jun 16 10:51:57 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 10:51:57 -0700 Subject: [Freeswitch-users] Voice lag in conference Message-ID: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/46b44ff5/attachment-0002.html From msc at freeswitch.org Tue Jun 16 11:35:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Jun 2009 11:35:52 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> Message-ID: <87f2f3b90906161135r195d5d7ch491e54440f78964e@mail.gmail.com> Can you describe your networking environment a bit? One thing that can affect the latency of your voice traffic is your network infrastructure. If you can isolate FS and some phones on a separate, controlled network then possibly you can start narrowing it down to other factors. -MC On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier wrote: > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/d443c49c/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 16 11:47:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Jun 2009 13:47:41 -0500 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> Message-ID: <191c3a030906161147l177abfbfq804ed5fb71a77afd@mail.gmail.com> The problem comes from the timing of certain phones during the capture of audio actually clocked slightly faster than what it advertises. Try the latest trunk with all the defaults in your sip profile as we have tried to make the defaults deal with this automatically. On Tue, Jun 16, 2009 at 12:51 PM, Bradley Brashier wrote: > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/9be60aa8/attachment-0002.html From bjbrashier at gmail.com Tue Jun 16 12:02:03 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 12:02:03 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <87f2f3b90906161135r195d5d7ch491e54440f78964e@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <87f2f3b90906161135r195d5d7ch491e54440f78964e@mail.gmail.com> Message-ID: <7bcfdd290906161202h78379859r9484a7aeed6e9877@mail.gmail.com> I have two different network setups, and have seen similar lag on both. The first is my home testbed. I'm connected to the internet through a home router and then a cablemodem. The home environment is pretty spare, of course. 2 machines and a couple of T-mobile cell phones with their SIP communication is all that goes through there. I have used my cell phone, a couple of different softphones, Gizmo call-ins, and regular PSTN calls. The worst lag is the T-mobile cell phones, but I'm happy to write that off as T-mobile's problem if we'd like. The second is the debug server environment on the systems where the conference product will eventually reside. The system is very complex, as it is already running a major hosted PBX service written years ago. I'm afraid all of the details of this system are beyond me, but I know that it includes a PSTN gateway, more T1s than I can count, and I'm having to split the RTP and SIP packets on separate ports for security and organizational purposes. For call-ins, I have used T-mobile again and regular PSTN, no softphones (yet). Obviously, this is the important environment, and the PSTN lag is somewhere around 500-700 ms (subjective). So am I correct in understanding that this is not a common issue, then, and that something can theoretically be done to help it? On Tue, Jun 16, 2009 at 11:35 AM, Michael Collins wrote: > Can you describe your networking environment a bit? One thing that can > affect the latency of your voice traffic is your network infrastructure. If > you can isolate FS and some phones on a separate, controlled network then > possibly you can start narrowing it down to other factors. > > -MC > > On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier > wrote: > >> I'm creating a conferencing product for use in a system with >> theoretically several hundred concurrent calls. I'm using FreeSwitch to >> create this product, but am not only new to FreeSwitch, but also the entire >> telecom industry as well as Open Source projects in general (I'm a >> recovering BIOS guy). >> >> I've got a bare-bones conference up and running on the server, including a >> handshake and a couple of features, and am using the default packages from >> the current trunk, but I've noticed that voice lag is a pretty big issue. >> Common lag times are several hundred milliseconds, and I've heard as long as >> a second. It seems to be at least marginally specific to individual phones >> -- certain phones have longer lag than others even on the same call. >> >> My question is really about what my options are. Is this just a part of >> SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim >> down that will help? Is this a common issue? If it's common, is it expected >> by the marketplace? >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/8064790e/attachment-0002.html From joshm at wabashcenter.com Tue Jun 16 12:01:13 2009 From: joshm at wabashcenter.com (Josh Moon) Date: Tue, 16 Jun 2009 15:01:13 -0400 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> Message-ID: <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn't a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/c8ce5a1c/attachment-0002.html From Prometheus001 at gmx.net Tue Jun 16 13:11:05 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 16 Jun 2009 22:11:05 +0200 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> <4A37DB1C.8070706@gmx.net> <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> Message-ID: <4A37FC59.60401@gmx.net> Thanks Michael, I have disabled it now. I finally got it to work, (sip_h_Call-Info=;answer-after=0) but the behaviour was not as desired, as I didn't manage the phone to pick up the call on the headset. It will only have the speaker enabled. So I will have to go a different way with parking the call and then forward it. Best regards Peter Michael Jerris schrieb: > uuid_setvar sip_invite_params intercom=true should be > unnecessary. > > Mike > > On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: > > >> It mainly works now by uuid_transfer the following way via event >> socket. >> uuid_setvar sip_invite_params intercom=true >> uuid_setvar sip_auto_answer true >> uuid_transfer 1000 XML default >> so the call is transferred from 1000 to 1000. >> >> What happens: >> 1) If I disable intercom on the Snom phone, the phone rings, stops >> ringing and rings again (ok) >> 1) If I enable intercom on the Snom phone, the phone rings, stops >> ringing and hangs up (not ok) >> >> So I do not get the Snom to pick up the call in intercom mode. >> >> The last invite is: >> INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib >> SIP/2.0 >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF >> Route: ;transport=tls;line=er6kxnib >> Max-Forwards: 68 >> From: "Peter FS" ;tag=9eQ8rjQy533HF >> To: >> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e >> CSeq: 116467629 INVITE >> Contact: >> Call-Info: ;answer-after=0 >> The intercom part is there and the Call-Info line with answer-after >> also. >> >> The phone answers with >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF >> From: "Peter FS" ;tag=9eQ8rjQy533HF >> To: >> > >>> ;tag=71rskygkr2 >>> >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e >> CSeq: 116467629 INVITE >> Contact: >> ;reg-id=1 >> WWW-Authenticate: Digest realm="sip2.mycompany.de", >> nonce="2ee26efe6ab27f88", algorithm=MD5 >> Content-Length: 0 >> and hangs up. >> >> Anybody know how to solve this Snom intercom issue? >> >> Best regards >> Peter >> >> >> Michael Jerris schrieb: >> >>> The transfer should work but it sounds like offhook agents is what >>> your really trying to accomplish so I would go with brian's >>> suggestion. >>> >>> >>> >>> On Jun 16, 2009, at 7:38 AM, Peter P GMX >>> wrote: >>> >>> >>> >>>> Hello Michael, >>>> >>>> I want the phone be ringing, just for acoustical feedback reasons. >>>> >>>> But what if I >>>> >>>> * transfer it to the same user destination again (now with >>>> intercom >>>> enabled), will this work? >>>> * transfer it to park and then transfer it to the same destination >>>> again (now with intercom enabled) >>>> >>>> Best regards >>>> Peter >>>> >>>> Michael Jerris schrieb: >>>> >>>> >>>>> The only way I can think to do this today would be to cancel the >>>>> call >>>>> and re send with the intercom headers for a phone that supports it. >>>>> It may be possible to send a reinvite with autoanswer headers but I >>>>> doubt that would work, all you could do is try making code to do it >>>>> it >>>>> a sipp or sipsak scenario and test it. A better aproach might be >>>>> to >>>>> answer the call normally and detect that to start your web workflow >>>>> or >>>>> not really ring the phone, just the web app and deliver the call >>>>> with >>>>> autoanswer when the button is hit in the web ui. >>>>> >>>>> Mike >>>>> >>>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX >>>>> wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Hello Brian, >>>>>> >>>>>> this is too easy :-). >>>>>> >>>>>> This is for a small callcenter app and I only want the user to >>>>>> pickup >>>>>> the call once (to accept the call in X-Lite (or a Snom phone) >>>>>> and to >>>>>> start the workflow on the web application). I do not want him to >>>>>> accept >>>>>> the call on the phone and then on the Web app. >>>>>> >>>>>> Best regards >>>>>> Peter >>>>>> >>>>>> >>>>>> >>>>>> Brian West schrieb: >>>>>> >>>>>> >>>>>> >>>>>>> click on the AA button? :) >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> What is the best way to have this done? Move the call to park >>>>>>>> and >>>>>>>> then >>>>>>>> retransfer again with intercom, or is there a better solution? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From bjbrashier at gmail.com Tue Jun 16 14:02:21 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 14:02:21 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> Message-ID: <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon wrote: > I am not as knowledgeable as the developers that will respond to your > question but I had the same problem as you. Here is what I did to combat > the delay: > > > > First off I started everything from scratch. I reinstalled Linux and then > I reinstalled FreeSWITCH by creating .deb packages. > > I then created my own conference profile and set the sample rate to 4000 > and changed the energy level to 20. > > I also made sure to test the conference room from phones that were in > completely different areas so there wasn?t a chance for feedback or really > bad echoing problems. > > > > Once I knew the delay was solved I raised the sample rate to 8000. I > tested it to make sure it would work properly. > > > > As Michael stated, this could be your network infrastructure but I just > wanted to let another FreeSWITCH user know what I did to try and stop the > voice delay. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley > Brashier > *Sent:* Tuesday, June 16, 2009 1:52 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Voice lag in conference > > > > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/35751cf0/attachment-0002.html From bjbrashier at gmail.com Tue Jun 16 14:26:33 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 14:26:33 -0700 Subject: [Freeswitch-users] Controlling Conference Controls Message-ID: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing "lock" and "unlock" actions and merge them into a "lock toggle" action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? There are a variety of small things like this that I need to implement. Would I be better off switching to Lua? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/7fc87dcf/attachment-0002.html From joshm at wabashcenter.com Tue Jun 16 14:30:56 2009 From: joshm at wabashcenter.com (Josh Moon) Date: Tue, 16 Jun 2009 17:30:56 -0400 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com><9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> Message-ID: <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> I was able to reduce it considerably. I can't say it is completely gone but I am very confident the ~.5 second delay I hear is because of the time it takes my voice to go through the leaps and bounds of the phone company to our server. I had at least a 3-5 second delay before I experimented with the conference settings. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 5:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Voice lag in conference I'm not sure I've got the opportunity to do that at the moment, but I do appreciate the point of view of a fellow product user. Were you able to eliminate noticeable lag, or just reduce it to reasonable levels? I'll try to do something similar when I update to the newest trunk as Anthony suggested. My copy is only a week old, but I'll try whatever has a chance of working, and I know you guys have been working on conferencing (the Moderator function couldn't have been timed better for me!). On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon > wrote: I am not as knowledgeable as the developers that will respond to your question but I had the same problem as you. Here is what I did to combat the delay: First off I started everything from scratch. I reinstalled Linux and then I reinstalled FreeSWITCH by creating .deb packages. I then created my own conference profile and set the sample rate to 4000 and changed the energy level to 20. I also made sure to test the conference room from phones that were in completely different areas so there wasn't a chance for feedback or really bad echoing problems. Once I knew the delay was solved I raised the sample rate to 8000. I tested it to make sure it would work properly. As Michael stated, this could be your network infrastructure but I just wanted to let another FreeSWITCH user know what I did to try and stop the voice delay. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bradley Brashier Sent: Tuesday, June 16, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Voice lag in conference I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/191d7951/attachment-0002.html From msc at freeswitch.org Tue Jun 16 14:38:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 16 Jun 2009 14:38:42 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> Message-ID: <87f2f3b90906161438v2246837fme8068aa13ea51240@mail.gmail.com> What is the big picture application? Reason I ask is that the FS devs and community have a lot of experience so if they can see the big picture they might be able to offer better advice. -MC On Tue, Jun 16, 2009 at 2:26 PM, Bradley Brashier wrote: > How much power do I have with DTMF conference controls? The wiki doesn't > have much information on this. For example, one of the things I'd like to do > is take the currently existing "lock" and "unlock" actions and merge them > into a "lock toggle" action. Preferably in XML configuration files. Is this > even possible? If so, how would I get started? > > There are a variety of small things like this that I need to implement. > Would I be better off switching to Lua? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/4d45e82e/attachment-0002.html From diego.viola at gmail.com Tue Jun 16 14:39:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 17:39:02 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? Message-ID: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego From intralanman at freeswitch.org Tue Jun 16 14:41:25 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 16 Jun 2009 17:41:25 -0400 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> Message-ID: <4A381185.9060806@freeswitch.org> Bradley Brashier wrote: > How much power do I have with DTMF conference controls? The wiki > doesn't have much information on this. For example, one of the things > I'd like to do is take the currently existing "lock" and "unlock" > actions and merge them into a "lock toggle" action. Preferably in XML > configuration files. Is this even possible? If so, how would I get > started? you could do this by having a script listen on the event socket... instead of using the default controls, you could just listen for a certain dtmf and then send the [un]lock command to the conference over the event socket -Ray From dule.maillist at gmail.com Tue Jun 16 14:43:28 2009 From: dule.maillist at gmail.com (Dan Le) Date: Tue, 16 Jun 2009 17:43:28 -0400 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24045824.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> Message-ID: <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> If you want FS server A to be able to call FS server B, you can set up a user account in server B's FS directory configs, and then just treat server B as a normal gateway by adding a gateway definition in server A. That will allow you to route calls to server B from A; to do the reverse, just mirror the configs the other direction. On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > > I like to connect two freeswitch, call each other, communicate and vice > versa. > Can you give me an example for that? > > Thanks > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/b2757e14/attachment-0002.html From pete at privateconnect.com Tue Jun 16 14:48:40 2009 From: pete at privateconnect.com (pete at privateconnect.com) Date: Tue, 16 Jun 2009 14:48:40 -0700 Subject: [Freeswitch-users] =?utf-8?q?Which_GSM_gateway_to_buy=3F?= Message-ID: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/e698cd74/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 16 14:55:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Jun 2009 16:55:39 -0500 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> Message-ID: <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> don't forget to read my suggestion too from earlier today =D On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon wrote: > I was able to reduce it considerably. I can?t say it is completely gone > but I am very confident the ~.5 second delay I hear is because of the time > it takes my voice to go through the leaps and bounds of the phone company to > our server. I had at least a 3-5 second delay before I experimented with > the conference settings. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley > Brashier > *Sent:* Tuesday, June 16, 2009 5:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Voice lag in conference > > > > I'm not sure I've got the opportunity to do that at the moment, but I do > appreciate the point of view of a fellow product user. Were you able to > eliminate noticeable lag, or just reduce it to reasonable levels? > > > > I'll try to do something similar when I update to the newest trunk as > Anthony suggested. My copy is only a week old, but I'll try whatever has a > chance of working, and I know you guys have been working on conferencing > (the Moderator function couldn't have been timed better for me!). > > On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon > wrote: > > I am not as knowledgeable as the developers that will respond to your > question but I had the same problem as you. Here is what I did to combat > the delay: > > > > First off I started everything from scratch. I reinstalled Linux and then > I reinstalled FreeSWITCH by creating .deb packages. > > I then created my own conference profile and set the sample rate to 4000 > and changed the energy level to 20. > > I also made sure to test the conference room from phones that were in > completely different areas so there wasn?t a chance for feedback or really > bad echoing problems. > > > > Once I knew the delay was solved I raised the sample rate to 8000. I > tested it to make sure it would work properly. > > > > As Michael stated, this could be your network infrastructure but I just > wanted to let another FreeSWITCH user know what I did to try and stop the > voice delay. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley > Brashier > *Sent:* Tuesday, June 16, 2009 1:52 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Voice lag in conference > > > > I'm creating a conferencing product for use in a system with theoretically > several hundred concurrent calls. I'm using FreeSwitch to create this > product, but am not only new to FreeSwitch, but also the entire telecom > industry as well as Open Source projects in general (I'm a recovering BIOS > guy). > > I've got a bare-bones conference up and running on the server, including a > handshake and a couple of features, and am using the default packages from > the current trunk, but I've noticed that voice lag is a pretty big issue. > Common lag times are several hundred milliseconds, and I've heard as long as > a second. It seems to be at least marginally specific to individual phones > -- certain phones have longer lag than others even on the same call. > > My question is really about what my options are. Is this just a part of > SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim > down that will help? Is this a common issue? If it's common, is it expected > by the marketplace? > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/df98338a/attachment-0002.html From bjbrashier at gmail.com Tue Jun 16 14:58:03 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 14:58:03 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> Message-ID: <7bcfdd290906161458n42e9479are572b462387f8adb@mail.gmail.com> Will do, just haven't had the time, yet! On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > don't forget to read my suggestion too from earlier today =D > > > > On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon wrote: > >> I was able to reduce it considerably. I can?t say it is completely gone >> but I am very confident the ~.5 second delay I hear is because of the time >> it takes my voice to go through the leaps and bounds of the phone company to >> our server. I had at least a 3-5 second delay before I experimented with >> the conference settings. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >> Brashier >> *Sent:* Tuesday, June 16, 2009 5:02 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Voice lag in conference >> >> >> >> I'm not sure I've got the opportunity to do that at the moment, but I do >> appreciate the point of view of a fellow product user. Were you able to >> eliminate noticeable lag, or just reduce it to reasonable levels? >> >> >> >> I'll try to do something similar when I update to the newest trunk as >> Anthony suggested. My copy is only a week old, but I'll try whatever has a >> chance of working, and I know you guys have been working on conferencing >> (the Moderator function couldn't have been timed better for me!). >> >> On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon >> wrote: >> >> I am not as knowledgeable as the developers that will respond to your >> question but I had the same problem as you. Here is what I did to combat >> the delay: >> >> >> >> First off I started everything from scratch. I reinstalled Linux and then >> I reinstalled FreeSWITCH by creating .deb packages. >> >> I then created my own conference profile and set the sample rate to 4000 >> and changed the energy level to 20. >> >> I also made sure to test the conference room from phones that were in >> completely different areas so there wasn?t a chance for feedback or really >> bad echoing problems. >> >> >> >> Once I knew the delay was solved I raised the sample rate to 8000. I >> tested it to make sure it would work properly. >> >> >> >> As Michael stated, this could be your network infrastructure but I just >> wanted to let another FreeSWITCH user know what I did to try and stop the >> voice delay. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >> Brashier >> *Sent:* Tuesday, June 16, 2009 1:52 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Voice lag in conference >> >> >> >> I'm creating a conferencing product for use in a system with theoretically >> several hundred concurrent calls. I'm using FreeSwitch to create this >> product, but am not only new to FreeSwitch, but also the entire telecom >> industry as well as Open Source projects in general (I'm a recovering BIOS >> guy). >> >> I've got a bare-bones conference up and running on the server, including a >> handshake and a couple of features, and am using the default packages from >> the current trunk, but I've noticed that voice lag is a pretty big issue. >> Common lag times are several hundred milliseconds, and I've heard as long as >> a second. It seems to be at least marginally specific to individual phones >> -- certain phones have longer lag than others even on the same call. >> >> My question is really about what my options are. Is this just a part of >> SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim >> down that will help? Is this a common issue? If it's common, is it expected >> by the marketplace? >> >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee you should not >> disseminate, distribute or copy this e-mail. Please notify the sender >> immediately by e-mail if you have received this e-mail by mistake and delete >> this e-mail from your system. E-mail transmission cannot be guaranteed to be >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender >> therefore does not accept liability for any errors or omissions in the >> contents of this message, which arise as a result of e-mail transmission. If >> verification is required please request a hard-copy version. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee you should not >> disseminate, distribute or copy this e-mail. Please notify the sender >> immediately by e-mail if you have received this e-mail by mistake and delete >> this e-mail from your system. E-mail transmission cannot be guaranteed to be >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender >> therefore does not accept liability for any errors or omissions in the >> contents of this message, which arise as a result of e-mail transmission. If >> verification is required please request a hard-copy version. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/3d05a52d/attachment-0002.html From jmesquita at gmail.com Tue Jun 16 15:01:22 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 16 Jun 2009 19:01:22 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> Message-ID: <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> Get Khomp GSM cars! Ihihihih They will soon be compatible with FreeSWITCH. Laterz, jmesquita On Tue, Jun 16, 2009 at 6:48 PM, wrote: > I did a fair amount of research into GSM gateways about 8 months ago. I > should first ask what are you looking to do with the gateway? > > -pete > > > -------- Original Message -------- > Subject: [Freeswitch-users] Which GSM gateway to buy? > From: Diego Viola > Date: Tue, June 16, 2009 2:39 pm > To: freeswitch-users at lists.freeswitch.org > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/062a6b65/attachment-0002.html From diego.viola at gmail.com Tue Jun 16 15:01:35 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 18:01:35 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> Message-ID: <86a32abc0906161501h2f5b634eq20de9eb2a58486ec@mail.gmail.com> I need it for gsm termination, I'd like to start with 8 channels, then 16, etc. Thanks, Diego On Tue, Jun 16, 2009 at 5:48 PM, wrote: > I did a fair amount of research into GSM gateways about 8 months ago.? I > should first ask what are you looking to do with the gateway? > > -pete > > -------- Original Message -------- > Subject: [Freeswitch-users] Which GSM gateway to buy? > From: Diego Viola > Date: Tue, June 16, 2009 2:39 pm > To: freeswitch-users at lists.freeswitch.org > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From edpimentl at gmail.com Tue Jun 16 15:09:35 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 16 Jun 2009 18:09:35 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> Message-ID: <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> For those that understand Portuguese http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a61e8051/attachment-0002.html From diego.viola at gmail.com Tue Jun 16 15:21:13 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 18:21:13 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> Message-ID: <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> We are a start-up company btw. On Tue, Jun 16, 2009 at 6:09 PM, EdPimentl wrote: > For those that understand Portuguese > http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve at justfone.com Tue Jun 16 15:21:06 2009 From: steve at justfone.com (Steven Brown) Date: Tue, 16 Jun 2009 23:21:06 +0100 Subject: [Freeswitch-users] Which GSM gateway to buy? Message-ID: <3e6d7b0c0906161521k204b9cawf19789d50fc92c98@mail.gmail.com> Hi , I have used PORTech single and double channel units on a couple of small projects with FS and they seem to have worked well in a low volume application . Have never tried one of the larger channel count ones yet for high call volumes though so cant verify how they perform, although just starting a larger project using 3 x 8 SIM PORTech units so will be able to give feedback on these in a few weeks. Steve Message: 2 Date: Tue, 16 Jun 2009 17:39:02 -0400 From: Diego Viola Subject: [Freeswitch-users] Which GSM gateway to buy? To: freeswitch-users at lists.freeswitch.org Message-ID: <86a32abc0906161439v89fbb58kcfe8297687dee600 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/b7524e13/attachment-0002.html From diego.viola at gmail.com Tue Jun 16 15:22:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 16 Jun 2009 18:22:47 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> Message-ID: <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> So we can't afford the top and the latest hardware. On Tue, Jun 16, 2009 at 6:21 PM, Diego Viola wrote: > We are a start-up company btw. > > On Tue, Jun 16, 2009 at 6:09 PM, EdPimentl wrote: >> For those that understand Portuguese >> http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/ >> -E >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From paul.degt at gmail.com Tue Jun 16 15:50:51 2009 From: paul.degt at gmail.com (paul.degt) Date: Tue, 16 Jun 2009 18:50:51 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> Message-ID: <4A3821CB.2070904@gmail.com> API CALL [global_getvar()] output: external_ssl_enable=false external_tls_port=5081 external_sip_port=5080 external_auth_calls=false internal_ssl_dir=/var/opt/freeswitch/conf/ssl internal_sip_port=5060 default_provider_contact=5000 default_provider_from_domain=example.com default_provider_password=password external_rtp_ip=74.92.196.241 xmpp_server_profile=xmpps xmpp_client_profile=xmppc global_codec_prefs=G722,PCMU,PCMA,GSM hold_music=local_stream://moh external_ssl_dir=/var/opt/freeswitch/conf/ssl internal_auth_calls=true local_ip_v4=192.168.0.40 unroll_loops=true default_areacode=918 default_provider_register=false local_mask_v4=255.255.255.0 default_password=1234 call_debug=false local_ip_v6=::1 default_provider_username=joeuser sound_prefix=/var/opt/freeswitch/sounds/en/us/callie outbound_caller_id=0000000000 default_country=US base_dir=/var/opt/freeswitch bind_server_ip=auto internal_tls_port=5061 switch_serial=c0a8002854db default_provider=example.com outbound_codec_prefs=PCMU,PCMA,GSM domain_name=192.168.0.40 domain=192.168.0.40 external_sip_ip=74.92.196.241 outbound_caller_name=Versafon.com rs-ring=%(1000, 4000, 425.0, 0.0) sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) internal_ssl_enable=false console_loglevel=debug uk-ring=%(400,200,400,450);%(400,2200,400,450) us-ring=%(2000, 4000, 440.0, 480.0) sip_tls_version=tlsv1 fr-ring=%(1500, 3500, 440.0, 0.0) bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) Brian West wrote: > Can you please put it back to auto-nat and email me the output of > global_getvar from the CLI so I can see what it detected? > > /b > > On Jun 16, 2009, at 7:18 AM, paul.degt at gmail.com wrote: > > >> Solved by replacing "auto-nat" with public ip in public profile >> "external_sip-ip" and "extrenal-rtp-ip" params. >> I believe values for these params used to be taken from vars.xml and >> so >> would have public ips by default - would be nice to document such >> changes in README. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve at justfone.com Tue Jun 16 15:57:53 2009 From: steve at justfone.com (Steven Brown) Date: Tue, 16 Jun 2009 23:57:53 +0100 Subject: [Freeswitch-users] How to delay audio ? Message-ID: <3e6d7b0c0906161557w4091d00ck4169ab952a671d56@mail.gmail.com> Hi All, I have a requirement to delay the audio sent from the calling channel in a call by a specified delay, much the same as the delay_echo functionality in the dptools but in a bridged rather than loopback mode. I cant immediately see a way to achieve this, is this something I'm missing or should I have look at adapting the delay_echo functionality. Thanks Steve Steven Brown email steve at justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. From anthony.minessale at gmail.com Tue Jun 16 16:22:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 16 Jun 2009 18:22:31 -0500 Subject: [Freeswitch-users] How to delay audio ? In-Reply-To: <3e6d7b0c0906161557w4091d00ck4169ab952a671d56@mail.gmail.com> References: <3e6d7b0c0906161557w4091d00ck4169ab952a671d56@mail.gmail.com> Message-ID: <191c3a030906161622k1b1444f6o350e043021539770@mail.gmail.com> if it's sip, turn on the jiterbuffer before you answer set the var jitterbuffer_msec=x where x is desired number of milliseconds (not too much!) On Tue, Jun 16, 2009 at 5:57 PM, Steven Brown wrote: > Hi All, > > I have a requirement to delay the audio sent from the calling channel > in a call by a specified delay, much the same as the delay_echo > functionality in the dptools but in a bridged rather than loopback > mode. I cant immediately see a way to achieve this, is this something > I'm missing or should I have look at adapting the delay_echo > functionality. > > Thanks > > Steve > > Steven Brown > > email steve at justfone.com > office 08707706968 > mobile 07768755409 > fax 07884636663 > > Justfone - Company Reg. No. : 3926817 > > Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 > 1EW > > The contents of this e-mail may be privileged and are confidential. It > may not be disclosed to or used by anyone other than the addressee(s), > nor copied in any way. If received in error, please advise sender, > then delete it from your system. Internet email communications are not > secure and therefore Justfone do not accept legal responsibility for > the contents of this message. Any views or opinions presented are > solely those of the author and do not necessarily represent those of > Justfone unless otherwise specifically stated. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/020f41fa/attachment-0002.html From bjbrashier at gmail.com Tue Jun 16 16:33:22 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 16 Jun 2009 16:33:22 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A381185.9060806@freeswitch.org> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> Message-ID: <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> Hmmm.... is that going to be easier than just modifying the mod_conference code to allow for a handfull of extra, simple commands? To me, it seems like for reasons of maintainability, etc, you want as few varied pieces as possible, in as few languages as possible. Socket scripting doesn't sound like it would be an extension of what I'm doing, now, more like a totally new method. Of course, I'm saying this from a complete outside point of view, and am more than willing to admit that I don't necessarily know the best course. On Tue, Jun 16, 2009 at 2:41 PM, Raymond Chandler < intralanman at freeswitch.org> wrote: > Bradley Brashier wrote: > > How much power do I have with DTMF conference controls? The wiki > > doesn't have much information on this. For example, one of the things > > I'd like to do is take the currently existing "lock" and "unlock" > > actions and merge them into a "lock toggle" action. Preferably in XML > > configuration files. Is this even possible? If so, how would I get > > started? > you could do this by having a script listen on the event socket... > instead of using the default controls, you could just listen for a > certain dtmf and then send the [un]lock command to the conference over > the event socket > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/4e83a6a2/attachment-0002.html From william.suffill at gmail.com Tue Jun 16 17:04:36 2009 From: william.suffill at gmail.com (William Suffill) Date: Tue, 16 Jun 2009 20:04:36 -0400 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> Message-ID: <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> It depends pretty heavily on what you are trying to add function wise. If it's more in depth using the event socket would allow it to be used on any FreeSwitch server assuming it caught the dtmf and acted according without having to modify the core source code/recompile. It might be a bit more work at first but could be well worth it depending on your needs. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/786acbf3/attachment-0002.html From evilla at chipoly.com Tue Jun 16 18:22:48 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Tue, 16 Jun 2009 19:22:48 -0600 Subject: [Freeswitch-users] MadBoss Conferences Examples - bug? Message-ID: <000601c9eeea$20a56d00$61f04700$@com> Hello friends. I've been playing with the mad boss examples. There is an issue I'd like to see: For example in MadBoss3: The first leg added to conference is the loopback/9999. Then you can add more users by conference_set_auto_outcall function. The problem I see is that: 1) Loopback music is still in the background of conference. 2) When everyone hang up, the conference is still active, because the 9999 user (music) is still inside the room. How can music be stoped once meeting is going to start? Edwin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/f8290b42/attachment-0002.html From darklion11 at yahoo.com Tue Jun 16 18:27:32 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:27:32 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> Message-ID: <24065535.post@talk.nabble.com> Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:28:21 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:28:21 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24065535.post@talk.nabble.com> Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:28:55 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:28:55 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24065535.post@talk.nabble.com> Actually my plan is if FS Server A has an account of 8011105, FS Server B shouldn't create another directory config. The user most not create an account 8011105 ON FS Server B. Single account for two servers. When I used a gateway config, yes its working but it needs a username and password My FS A = 192.168.0.104 My FS B = 192.168.0.105 My sample sip_profiles/external/gwfsa.xml I log as 8011104 and call 8011107 When I used this config on FS Server A and I called to FS B (8011107) the caller user id is 8011105 and the ip is 192.168.0.104 Is there another way to manage the gateway with the caller id of the user not the gateway user id and is there a gateway that doesn't need a username and password? Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24065535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:42:15 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:42:15 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: References: <24045890.post@talk.nabble.com> Message-ID: <24065638.post@talk.nabble.com> my nibble.conf.xml Account 1001.xml I check unixodbc has been installed. # isql zenoss edmar edmar [SQL]> Connected successfully but on freeswitch error Cannot connect to user ODBC [root] Darren Schreiber wrote: > > What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the > real logs from FS's logs? The info below is not nearly detailed enough. > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Monday, June 15, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC > > > Hi > > I experiencing an error on mod_nibblebill. I already load it from > autoload_configs, especially mod_spidermonkey. Uncomment > mod_spidermonkey_odbc. I also download unixodbc and created the files > /etc/odbcinst.ini and /etc/odbc.ini with the correct format > > [zenoss] > DATABASE = tcapi > USER = root > PASS = password > ..... > > I type also on the console isql zenoss root password. Also working... > > But an error occur on freeswitch Cannot connect to user [root] ... > > What do you thinks is the problem? > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24045890.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24065638.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 18:43:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 18:43:59 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC Message-ID: <24065638.post@talk.nabble.com> my nibble.conf.xml param name="db_username" value="edmar"/> param name="db_password" value="edmar"/> param name="db_dsn" value="tcapi"/> param name="db_column_cash" value="cash"/> param name="db_column_account" value="id"/> param name="global_heartbeat" value="1"/> !-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. --> param name="lowbal_amt" value="5"/> param name="lowbal_action" value="play ding"/> param name="nobal_amt" value="0"/> param name="nobal_action" value="hangup"/> param name="percall_max_amt" value="100"/> param name="percall_action" value="hangup"/> Account 1001.xml param name="password" value="1234"/> param name="vm-password" value="1001"/> param name="vm-mailto" value=""/> param name="vm-email-all-messages" value="false"/> param name="vm-delete-file" value="false"/> param name="vm-attach-file" value="false"/> I check unixodbc has been installed. # isql zenoss edmar edmar [SQL]> Connected successfully but on freeswitch error Cannot connect to user ODBC [root] Darren Schreiber wrote: > > What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the > real logs from FS's logs? The info below is not nearly detailed enough. > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Monday, June 15, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC > > > Hi > > I experiencing an error on mod_nibblebill. I already load it from > autoload_configs, especially mod_spidermonkey. Uncomment > mod_spidermonkey_odbc. I also download unixodbc and created the files > /etc/odbcinst.ini and /etc/odbc.ini with the correct format > > [zenoss] > DATABASE = tcapi > USER = root > PASS = password > ..... > > I type also on the console isql zenoss root password. Also working... > > But an error occur on freeswitch Cannot connect to user [root] ... > > What do you thinks is the problem? > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24045890.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24065638.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at gmail.com Tue Jun 16 18:45:07 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 16 Jun 2009 22:45:07 -0300 Subject: [Freeswitch-users] MadBoss Conferences Examples - bug? In-Reply-To: <000601c9eeea$20a56d00$61f04700$@com> References: <000601c9eeea$20a56d00$61f04700$@com> Message-ID: <5a8712120906161845t228d192eo80b52c38f65bdaf8@mail.gmail.com> Look at the newly implemented wait-mod conference flag on mod_conference. This is: http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E under parameters->conference-flags jmesquita On Tue, Jun 16, 2009 at 10:22 PM, Ing. Edwin Villarreal wrote: > Hello friends. > > > > I?ve been playing with the mad boss examples. There is an issue I?d like > to see: > > > > For example in MadBoss3: > > The first leg added to conference is the loopback/9999? Then you can add > more users by conference_set_auto_outcall function. > > > > The problem I see is that: > > 1) Loopback music is still in the background of conference. > > 2) When everyone hang up, the conference is still active, because the > 9999 user (music) is still inside the room. > > > > How can music be stoped once meeting is going to start? > > > > *Edwin* > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/911fff4c/attachment-0002.html From cesar.bermudez at gmail.com Tue Jun 16 19:11:05 2009 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 16 Jun 2009 23:11:05 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> Message-ID: Diego, i'have a customer using 3 portech using todo termination on argentina with asterisk on high volume calls and they are working great. Best regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/c86d3f46/attachment-0002.html From edpimentl at gmail.com Tue Jun 16 19:34:27 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 16 Jun 2009 22:34:27 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: References: <20090616144840.2ad02225396a31c9de30536f2e338977.14268f62a7.wbe@email04.secureserver.net> <5a8712120906161501y4d08d764k14e0aaa19c5f2af0@mail.gmail.com> <9dc4a1670906161509o623326f3v69ef441d43443f38@mail.gmail.com> <86a32abc0906161521p57cb9e1dv79547274508f05a9@mail.gmail.com> <86a32abc0906161522q49a4b87p35cbb4cfe833a015@mail.gmail.com> Message-ID: <9dc4a1670906161934j24f2fcfej93d0286a0473fe0@mail.gmail.com> I have been using Portech for over two years and they work fine. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/f9d3a1a0/attachment-0002.html From brian at freeswitch.org Tue Jun 16 19:49:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 21:49:58 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24065535.post@talk.nabble.com> References: <24065535.post@talk.nabble.com> Message-ID: Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > Is there another way to manage the gateway with the caller id of the > user > not the gateway user id and is there a gateway that doesn't need a > username > and password? From evilla at chipoly.com Tue Jun 16 19:58:20 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Tue, 16 Jun 2009 20:58:20 -0600 Subject: [Freeswitch-users] [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() Message-ID: <003501c9eef7$79330790$6b9916b0$@com> Hello! I need some fresh ideas about this issue. My gateway is already REGED, but when REG expires and sofia is trying to renew REG, then it fails to register. . 2009-06-16 16:46:39 [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() chipoly Registration Failed with status DNS Error [503]. failure #14 . 2009-06-16 16:46:40 [WARNING] sofia_reg.c:334 sofia_reg_check_gateway() chipoly Failed Registration, setting retry to 450 seconds. Here is a complete before/after http://pastebin.freeswitch.org/9406 when doing sofia profile external restart, gateway REGs again, so it's not DNS problem. (I think) Thank you for ur help! Edwin Villarreal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/333eeaa7/attachment-0002.html From brian at freeswitch.org Tue Jun 16 20:03:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:03:05 -0500 Subject: [Freeswitch-users] [ERR] sofia_reg.c:1381 sofia_reg_handle_sip_r_register() In-Reply-To: <003501c9eef7$79330790$6b9916b0$@com> References: <003501c9eef7$79330790$6b9916b0$@com> Message-ID: <16C354B9-9744-4BFB-8DDD-A793A96D9E1A@freeswitch.org> This should be a huge clue... what might be your providers name? Seems something is missing here or you have the settings wrong. /b On Jun 16, 2009, at 9:58 PM, Ing. Edwin Villarreal wrote: > DNS Error [503]. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/32af6f9c/attachment-0002.html From darklion11 at yahoo.com Tue Jun 16 20:05:22 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 20:05:22 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> Message-ID: <24066210.post@talk.nabble.com> How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 On 192.168.0.4 Brian West-3 wrote: > > Turn off authentication or use ACL's > > /b > > On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > >> Is there another way to manage the gateway with the caller id of the >> user >> not the gateway user id and is there a gateway that doesn't need a >> username >> and password? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066210.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jonny.voip at gmail.com Tue Jun 16 20:05:54 2009 From: jonny.voip at gmail.com (Jonathan DiVita) Date: Tue, 16 Jun 2009 23:05:54 -0400 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 Message-ID: Hello, all. I'm currently playing around with a new install of Freeswitch and wanted to try out mod_opal. Below are the current SVN builds for opal, ptlib, and freeswitch. I end up with the following errors when compiling. making all mod_opal Compiling mod_opal.cpp... Compiling mod_opal.cpp ... In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ?virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)? /usr/include/opal/opal/localep.h:267: error: overriding ?virtual ptlib_virtual_function_changed_or_removed****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)?: mod_opal.cpp:564: error: no matching function for call to ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)? /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) make[4]: *** [mod_opal.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opal-all] Error 1 make[1]: *** [mod_opal] Error 2 make: *** [mod_opal] Error 2 root at Freeswitch1:~/opal# svn info Path: . URL: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/trunk Repository Root: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip Repository UUID: 023b2edf-31b2-4de3-b41e-bca80c47788f Revision: 22909 Node Kind: directory Schedule: normal Last Changed Author: rjongbloed Last Changed Rev: 22909 Last Changed Date: 2009-06-16 07:09:41 -0400 (Tue, 16 Jun 2009) root at Freeswitch1:~/opal# cd .. root at Freeswitch1:~# cd ptlib/ root at Freeswitch1:~/ptlib# svn info Path: . URL: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunk Repository Root: https://opalvoip.svn.sourceforge.net/svnroot/opalvoip Repository UUID: 023b2edf-31b2-4de3-b41e-bca80c47788f Revision: 22909 Node Kind: directory Schedule: normal Last Changed Author: csoutheren Last Changed Rev: 22907 Last Changed Date: 2009-06-16 05:49:19 -0400 (Tue, 16 Jun 2009) root at Freeswitch1:~/ptlib# cd /freeswitch/ root at Freeswitch1:/freeswitch# svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 13798 Node Kind: directory Schedule: normal Last Changed Author: brian Last Changed Rev: 13798 Last Changed Date: 2009-06-16 19:11:45 -0400 (Tue, 16 Jun 2009) Do I need earlier versions of opal and ptlib? Thanks! Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/87ecc74a/attachment-0002.html From brian at freeswitch.org Tue Jun 16 20:12:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:12:01 -0500 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 In-Reply-To: References: Message-ID: please see MODOPAL-10 on jira. /b On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > Hello, all. I'm currently playing around with a new install of > Freeswitch and wanted to try out mod_opal. Below are the current > SVN builds for opal, ptlib, and freeswitch. I end up with the > following errors when compiling. > > making all mod_opal > Compiling mod_opal.cpp... > Compiling mod_opal.cpp ... > In file included from mod_opal.cpp:25: > mod_opal.h:151: error: conflicting return type specified for > ?virtual OpalLocalConnection* > FSEndPoint::CreateConnection(OpalCall&, void*)? > /usr/include/opal/opal/localep.h:267: error: overriding ?virtual > ptlib_virtual_function_changed_or_removed****** > OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? > mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, > FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, > switch_channel_t*)?: > mod_opal.cpp:564: error: no matching function for call to > ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, > NULL)? > /usr/include/opal/opal/localep.h:290: note: candidates are: > OpalLocalConnection::OpalLocalConnection(OpalCall&, > OpalLocalEndPoint&, void*, unsigned int, > OpalConnection::StringOptions*, char) > /usr/include/opal/opal/localep.h:276: note: > OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) > make[4]: *** [mod_opal.lo] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_opal-all] Error 1 > make[1]: *** [mod_opal] Error 2 > make: *** [mod_opal] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment-0002.html From darklion11 at yahoo.com Tue Jun 16 20:03:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24065535.post@talk.nabble.com> Message-ID: <929908.28142.qm@web57310.mail.re1.yahoo.com> How can i turn off authentication? This is my acl.conf.xml on 192.168.0.105 On 192.168.0.4 ________________________________ From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 16, 2009 10:49:58 PM Subject: Re: [Freeswitch-users] How can I join two freeswitch on two servers? Turn off authentication or use ACL's /b On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > Is there another way to manage the gateway with the caller id of the > user > not the gateway user id and is there a gateway that doesn't need a > username > and password? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/2839a86d/attachment-0002.html From brian at freeswitch.org Tue Jun 16 20:31:21 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:31:21 -0500 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A3821CB.2070904@gmail.com> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> Message-ID: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> Shouldn't have really changed any behavior at all... What svn rev are you on? /b On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > API CALL [global_getvar()] output: > external_ssl_enable=false > external_tls_port=5081 > external_sip_port=5080 > external_auth_calls=false > internal_ssl_dir=/var/opt/freeswitch/conf/ssl > internal_sip_port=5060 > default_provider_contact=5000 > default_provider_from_domain=example.com > default_provider_password=password > external_rtp_ip=74.92.196.241 > xmpp_server_profile=xmpps > xmpp_client_profile=xmppc > global_codec_prefs=G722,PCMU,PCMA,GSM > hold_music=local_stream://moh > external_ssl_dir=/var/opt/freeswitch/conf/ssl > internal_auth_calls=true > local_ip_v4=192.168.0.40 > unroll_loops=true > default_areacode=918 > default_provider_register=false > local_mask_v4=255.255.255.0 > default_password=1234 > call_debug=false > local_ip_v6=::1 > default_provider_username=joeuser > sound_prefix=/var/opt/freeswitch/sounds/en/us/callie > outbound_caller_id=0000000000 > default_country=US > base_dir=/var/opt/freeswitch > bind_server_ip=auto > internal_tls_port=5061 > switch_serial=c0a8002854db > default_provider=example.com > outbound_codec_prefs=PCMU,PCMA,GSM > domain_name=192.168.0.40 > domain=192.168.0.40 > external_sip_ip=74.92.196.241 > outbound_caller_name=Versafon.com > rs-ring=%(1000, 4000, 425.0, 0.0) > sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) > internal_ssl_enable=false > console_loglevel=debug > uk-ring=%(400,200,400,450);%(400,2200,400,450) > us-ring=%(2000, 4000, 440.0, 480.0) > sip_tls_version=tlsv1 > fr-ring=%(1500, 3500, 440.0, 0.0) > bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) > From brian at freeswitch.org Tue Jun 16 20:36:38 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 22:36:38 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <929908.28142.qm@web57310.mail.re1.yahoo.com> References: <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> Message-ID: Now you have to tell the sofia profile to use that ACL /b On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > How can i turn off authentication? This is my acl.conf.xml on > 192.168.0.105 > > > > > > On 192.168.0.4 > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/47e21916/attachment-0002.html From darklion11 at yahoo.com Tue Jun 16 21:00:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:00:04 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> Message-ID: <24066647.post@talk.nabble.com> How can sofia profile can call ACL? Can you give me an example? Brian West-3 wrote: > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 21:02:21 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:02:21 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? Message-ID: <24066647.post@talk.nabble.com> How can sofia profile can call ACL? Can you give me an example? Like this? I put this on external profile "/> "/> Brian West-3 wrote: > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 16 21:08:46 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 23:08:46 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066647.post@talk.nabble.com> References: <24066647.post@talk.nabble.com> Message-ID: <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> COPY paste fail :) something like that as per the example. /b On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > > How can sofia profile can call ACL? > Can you give me an example? > Like this? > > I put this on external profile > > "/> > "/> > > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From darklion11 at yahoo.com Tue Jun 16 21:26:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:26:38 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> Message-ID: <24066825.post@talk.nabble.com> If FS A has an account 8011105 does FS B also nid to register 8011105? Yes it working on a gateway but the username of the gateway was shown on my softphone and also it nids a password for the gateway... is there an option to view the caller name and number of the FS A gateway to FS B? Brian West-3 wrote: > > COPY paste fail :) > > > > something like that as per the example. > > /b > > On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > >> >> How can sofia profile can call ACL? >> Can you give me an example? >> Like this? >> >> I put this on external profile >> >> "/> >> "/> >> >> >> Brian West-3 wrote: >>> >>> Now you have to tell the sofia profile to use that ACL >>> >>> /b >>> >>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>> >>>> How can i turn off authentication? This is my acl.conf.xml on >>>> 192.168.0.105 >>>> >>>> >>>> >>>> >>>> >>>> On 192.168.0.4 >>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 21:30:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 21:30:38 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066825.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> Message-ID: <24066849.post@talk.nabble.com> Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on FS B Edmar Cruz wrote: > > If FS A has an account 8011105 does FS B also nid to register 8011105? Yes > it working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066849.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Jun 16 21:33:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 16 Jun 2009 23:33:56 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066849.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24066849.post@talk.nabble.com> Message-ID: Its not an error its a warning and you don't have your ACL's configured correctly. You're trying too hard! :) set auth- calls=false on the profile. /b On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote: > > Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on > FS B From bipin at xbipin.com Tue Jun 16 22:00:59 2009 From: bipin at xbipin.com (xbipin) Date: Tue, 16 Jun 2009 22:00:59 -0700 (PDT) Subject: [Freeswitch-users] is there any available gui for freeswitch using cake php? In-Reply-To: <86a32abc0906160301p802a71arb8d67bc9a0ebcfa9@mail.gmail.com> References: <24031900.post@talk.nabble.com> <24032171.post@talk.nabble.com> <86a32abc0906151956q714d94b8u3d2c0c31982de8ed@mail.gmail.com> <24046873.post@talk.nabble.com> <86a32abc0906160217u79784073jea412766c4a7f97a@mail.gmail.com> <24050713.post@talk.nabble.com> <86a32abc0906160301p802a71arb8d67bc9a0ebcfa9@mail.gmail.com> Message-ID: <24067052.post@talk.nabble.com> hi, if u need any help, i can always provide that. Regards, Bipin Diego Viola wrote: > > Sure, I will let you know when it's done. > > On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruz wrote: >> >> Thanks for that info... Can you send me this project if and only if it is >> already finished on this email darklion at yahoo.com? Thanks a lot... >> >> >> Diego Viola wrote: >>> >>> I'm currently rewriting the entire thing, it was a commercial app >>> first, but I'm re-writing it in order to make it open source. It's not >>> ready yet, as soon as I finish it, I will release it to the public. >>> >>> Diego >>> >>> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz >>> wrote: >>>> >>>> Can you share me the link of it so i can try... Please >>>> >>>> Diego Viola wrote: >>>>> >>>>> I'm currently writing a rails app that uses mod_nibblebill for >>>>> billing, >>>>> it's >>>>> a calling card app. >>>>> >>>>> Diego >>>>> >>>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz >>>>> wrote: >>>>> >>>>>> >>>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet >>>>>> fully >>>>>> developed... >>>>>> Is there any GUI with billing options? >>>>>> >>>>>> >>>>>> seven-8 wrote: >>>>>> > >>>>>> > http://www.tcapi.org/index.php?title=Main_Page >>>>>> > >>>>>> > >>>>>> > >>>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote: >>>>>> > >>>>>> >> >>>>>> >> is there any available gui for freeswitch using cake php complete >>>>>> >> instead of >>>>>> >> wikipbx, spice softphone or pfsense? >>>>>> >> -- >>>>>> >> View this message in context: >>>>>> >> >>>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html >>>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>> >> >>>>>> >> >>>>>> >> _______________________________________________ >>>>>> >> Freeswitch-users mailing list >>>>>> >> Freeswitch-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Freeswitch-users mailing list >>>>>> > Freeswitch-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24032171.html >>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24046873.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24050713.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24067052.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From paul.degt at gmail.com Tue Jun 16 22:00:36 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Wed, 17 Jun 2009 01:00:36 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> Message-ID: <4A387874.5090105@gmail.com> 13564 Brian West wrote: > Shouldn't have really changed any behavior at all... What svn rev are > you on? > > /b > > On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > > >> API CALL [global_getvar()] output: >> external_ssl_enable=false >> external_tls_port=5081 >> external_sip_port=5080 >> external_auth_calls=false >> internal_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_sip_port=5060 >> default_provider_contact=5000 >> default_provider_from_domain=example.com >> default_provider_password=password >> external_rtp_ip=74.92.196.241 >> xmpp_server_profile=xmpps >> xmpp_client_profile=xmppc >> global_codec_prefs=G722,PCMU,PCMA,GSM >> hold_music=local_stream://moh >> external_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_auth_calls=true >> local_ip_v4=192.168.0.40 >> unroll_loops=true >> default_areacode=918 >> default_provider_register=false >> local_mask_v4=255.255.255.0 >> default_password=1234 >> call_debug=false >> local_ip_v6=::1 >> default_provider_username=joeuser >> sound_prefix=/var/opt/freeswitch/sounds/en/us/callie >> outbound_caller_id=0000000000 >> default_country=US >> base_dir=/var/opt/freeswitch >> bind_server_ip=auto >> internal_tls_port=5061 >> switch_serial=c0a8002854db >> default_provider=example.com >> outbound_codec_prefs=PCMU,PCMA,GSM >> domain_name=192.168.0.40 >> domain=192.168.0.40 >> external_sip_ip=74.92.196.241 >> outbound_caller_name=Versafon.com >> rs-ring=%(1000, 4000, 425.0, 0.0) >> sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) >> internal_ssl_enable=false >> console_loglevel=debug >> uk-ring=%(400,200,400,450);%(400,2200,400,450) >> us-ring=%(2000, 4000, 440.0, 480.0) >> sip_tls_version=tlsv1 >> fr-ring=%(1500, 3500, 440.0, 0.0) >> bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Tue Jun 16 22:06:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 00:06:30 -0500 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: <4A387874.5090105@gmail.com> References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> <4A387874.5090105@gmail.com> Message-ID: can you update and try that again? /b On Jun 17, 2009, at 12:00 AM, paul.degt at gmail.com wrote: > 13564 From darklion11 at yahoo.com Tue Jun 16 22:24:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 22:24:31 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24066849.post@talk.nabble.com> Message-ID: <24067204.post@talk.nabble.com> Yes its already set to false... What should I do? Brian West-3 wrote: > > Its not an error its a warning and you don't have your ACL's > configured correctly. You're trying too hard! :) set auth- > calls=false on the profile. > > /b > > On Jun 16, 2009, at 11:30 PM, Edmar Cruz wrote: > >> >> Error on freeswitch Can't find user [8011105 @192.168.0.105] ... on >> FS B > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24067204.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Tue Jun 16 23:11:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 16 Jun 2009 23:11:31 -0700 (PDT) Subject: [Freeswitch-users] Allow invites from another sip server? Message-ID: <24067552.post@talk.nabble.com> On my acl.conf.xml I allow the ip 116.50.110.2 Is this correct? Error sip_invite() ... Error occur rejected by acl domains param name="apply-inbound-acl" value="domains"/> param name="apply-register_acl" value="domains"/> -- View this message in context: http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From panselva at gmail.com Tue Jun 16 23:14:13 2009 From: panselva at gmail.com (selva kumar) Date: Wed, 17 Jun 2009 11:44:13 +0530 Subject: [Freeswitch-users] outbound error log Message-ID: <45f609f90906162314q3b450f4aid4d98c81af9a1e8f@mail.gmail.com> Hi Michael, I have pasted the freeswitch logs as requested in ( pastebin.freeswitch.org) Thanks Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/0a7867c8/attachment-0002.html From enno.egbert at web.de Tue Jun 16 23:18:45 2009 From: enno.egbert at web.de (NOx-WHV) Date: Tue, 16 Jun 2009 23:18:45 -0700 (PDT) Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> Message-ID: <24067617.post@talk.nabble.com> Hi, look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" where you can ask for special requirements. NOx Diego Viola wrote: > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Tue Jun 16 23:25:19 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 17 Jun 2009 08:25:19 +0200 Subject: [Freeswitch-users] Allow invites from another sip server? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E1E3@cooper> You need to change the apply-inbound-acl to your list (myip) instead of using the domains list. /Peter ----- Ursprungligt meddelande ----- Fr?n: Edmar Cruz Skickat: den 17 juni 2009 08:20 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Allow invites from another sip server? On my acl.conf.xml I allow the ip 116.50.110.2 Is this correct? Error sip_invite() ... Error occur rejected by acl domains param name="apply-inbound-acl" value="domains"/> param name="apply-register_acl" value="domains"/> -- View this message in context: http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a388b4b32938939812364! From grevenx at me.com Tue Jun 16 23:42:12 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 17 Jun 2009 08:42:12 +0200 Subject: [Freeswitch-users] Allow invites from another sip server? In-Reply-To: <24067552.post@talk.nabble.com> References: <24067552.post@talk.nabble.com> Message-ID: I would recomend you to go read on this page: http://wiki.freeswitch.org/wiki/Acl The short answer: replace "domains" with the name of the list you have actually created, that is "myip"... -- Best regards, Even Andr? On 17. juni. 2009, at 08.11, Edmar Cruz wrote: > > On my acl.conf.xml > > I allow the ip 116.50.110.2 > > Is this correct? > > > > > > Error sip_invite() ... Error occur rejected by acl domains > > param name="apply-inbound-acl" value="domains"/> > param name="apply-register_acl" value="domains"/> > > -- > View this message in context: http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yudha2008 at gmail.com Tue Jun 16 23:44:47 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 17 Jun 2009 12:14:47 +0530 Subject: [Freeswitch-users] javascript session.execute Message-ID: *Hi, I have some questions can any one assist me and answer this query Question 1: Can we able to execute all the api command through JavaScript using session.execute. Question 2: How to kill the session using uuid_kill whether it is possible. **If yes means how we will use uuid_kill in session* * i have tried some thing like this : session.execute("uuid_kill", "session.uuid"); session.execute("uuid_kill session.uuid"); but there is no output for the session.execute. Whether it is possible to execute uuid_kill in javascript Thanks in advance Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/4ba25ab3/attachment-0002.html From mcampbellsmith at gmail.com Tue Jun 16 23:58:23 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 17 Jun 2009 16:58:23 +1000 Subject: [Freeswitch-users] Porta Billing? Message-ID: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Hi! Does freeswitch support extracting the billing data (PortaBilling) in SIP messages? If so, is there anyway I can get that information to an extension? 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis? playMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4? bK1FF90 From: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=47E011-580 To: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=adfde4bc91cd85e752cb0672816ac1? a6-eb1b Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB? 7 CSeq: 3 REGISTER PortaBilling: available-funds:7.37 currency:AUD Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060};? expires=3595 Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 Thanks! From jason at jasonjgw.net Wed Jun 17 00:00:19 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 17:00:19 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue Message-ID: <20090617070019.GA5042@jdc.jasonjgw.net> this is an issue which I've been discussing with Brian West on IRC and in e-mail correspondence, which I thought I should bring to the list so that others can look at it as well. The configuration My external SIP profile has its ext-sip-ip and ext-rtp-ip set to stun:stun.freeswitch.org. This is necessary for nat traversal. I have an internal-ipv6 profile as well, which is working, but for some reason it's interfering with calls on the external profile (which of course is an IPv4 profile). The symptom is the following line in outgoing SIP messages while attempting to establish a call to a gateway via the external profile: o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org in response to which, the other side returns a 488 (invalid session description). I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the internal-ipv6.xml profile, but they are said as explained above in the external profile, where the problem lies. This looks like a bug to me but I want to rule out misconfiguration first. This has been tested on revision 13806. From panselva at gmail.com Wed Jun 17 00:02:26 2009 From: panselva at gmail.com (selva kumar) Date: Wed, 17 Jun 2009 12:32:26 +0530 Subject: [Freeswitch-users] Outboubd is not working while inbound is configured and runs well Message-ID: <45f609f90906170002k3938e307ha7f7ab3dd90418de@mail.gmail.com> Hi, I configured oubound in FS, it worked fine. Then I configured inbound in FS,it also worked fine.But now the inbound works fine and the outbound is not working. What is the reason? I got this error logs 2009-06-17 12:11:59 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/1005 at 172.20.192.153[d08bb613-5f40-40c1-8e68-f370177e34a0] 2009-06-17 12:11:59 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing aspro->876159840881185 in context public 2009-06-17 12:11:59 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/internal/1005 at 172.20.192.153[CS_EXECUTE] [NORMAL_CLEARING] 2009-06-17 12:11:59 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 60 (sofia/internal/1005 at 172.20.192.153) Ended 2009-06-17 12:11:59 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/ 1005 at 172.20.192.153 [CS_HANGUP] 2009-06-17 12:16:51 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/9374530072 at 172.20.191.228[d351bf10-2cb9-4cbc-a0fd-66843dc6342b] Can somebody help me on this. -- with regrds Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/196fbc9a/attachment-0002.html From jason at jasonjgw.net Wed Jun 17 00:15:19 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 17:15:19 +1000 Subject: [Freeswitch-users] Outboubd is not working while inbound is configured and runs well In-Reply-To: <45f609f90906170002k3938e307ha7f7ab3dd90418de@mail.gmail.com> References: <45f609f90906170002k3938e307ha7f7ab3dd90418de@mail.gmail.com> Message-ID: <20090617071519.GA9441@jdc.jasonjgw.net> selva kumar wrote: > Hi, > I configured oubound in FS, it worked fine. > Then I configured inbound in FS,it also worked fine.But now the inbound > works fine and the outbound is not working. > What is the reason? If you turn on debug-level logging, it might be possible to work out what is going on. Either run /log debug from fs_cli or make sure that the log level set in your configuration is debug, then capture the logs from your log files or fs_cli and examine the results to find out what is going on. From yivzhenko at mksat.net Wed Jun 17 00:24:15 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Wed, 17 Jun 2009 10:24:15 +0300 Subject: [Freeswitch-users] mod_nibblebill not set variablenibble_total_billed In-Reply-To: References: <200906091126.03556.yivzhenko@mksat.net> Message-ID: <200906171024.16279.yivzhenko@mksat.net> Ok, i understand! This variable sets for event CHANNEL_HANGUP_COMPLETE, not for CHANNEL_HANGUP :-) On Tuesday 16 June 2009 19:59:28 you wrote: > That should not be the case - I will double check this. My apologies if I > broke it. :-( > > Please file a bug on this so I don't forget. > > _____ > > From: Yuriy Ivzhenko [mailto:yivzhenko at mksat.net] > Sent: Tuesday, June 09, 2009 1:26 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] mod_nibblebill not set > variablenibble_total_billed > > > > Some time ago mod_nibblebill was set variable nibble_total_billed after > hangup. > > But after last few updates of module this variable is no more sets. > > Somebody else have this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/61c0b326/attachment-0002.html From grevenx at me.com Wed Jun 17 00:26:36 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 17 Jun 2009 09:26:36 +0200 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> References: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Message-ID: <513BEDE8-72A4-4B81-B1F3-6343F13DBB9B@me.com> I'm not sure if you are able to read that header from FS without any modifications. What you can try is the generic syntax to get header variables: {$sip_h_PortaBilling}. Not sure, but I think this might only work with a set of standard headers, as well as all non-standard headers, that really should be prefixed with "X-". If this does not work, the PortaBilling header should in this case be renamed to X-PortaBilling, and you would be able to get it from FreeSWITCH with: {$sip_h_X-PortaBilling}. That is if it's possible to change that header on the server that sends it... Best regards, Even Andr? On 17. juni. 2009, at 08.58, Mark Campbell-Smith wrote: > Hi! > > Does freeswitch support extracting the billing data (PortaBilling) in > SIP messages? If so, is there anyway I can get that information to an > extension? > > 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis playMsg: > Received: > SIP/2.0 200 OK > Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4 bK1FF90 > From: {sip:61XXXXXXXXX at sip.pennytel.com}; tag=47E011-580 > To: {sip:61XXXXXXXXX at sip.pennytel.com}; > tag=adfde4bc91cd85e752cb0672816ac1 a6-eb1b > Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB 7 > CSeq: 3 REGISTER > PortaBilling: available-funds:7.37 currency:AUD > Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060}; expires=3595 > Server: Sip EXpress router (0.9.6 (i386/freebsd)) > Content-Length: 0 > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at suspicious.org Wed Jun 17 00:32:03 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 17 Jun 2009 02:32:03 -0500 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Message-ID: I doubt that header is exposed since its not a standard sip header. However you could probably patch mod_sofia to expose it without too much trouble... How difficult that would be is dependant on where in session that comes in > From: Mark Campbell-Smith > Reply-To: > Date: Wed, 17 Jun 2009 16:58:23 +1000 > To: > Subject: [Freeswitch-users] Porta Billing? > > Hi! > > Does freeswitch support extracting the billing data (PortaBilling) in > SIP messages? If so, is there anyway I can get that information to an > extension? > > 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis? playMsg: > Received: > SIP/2.0 200 OK > Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4? bK1FF90 > From: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=47E011-580 > To: {sip:61XXXXXXXXX at sip.pennytel.com};? > tag=adfde4bc91cd85e752cb0672816ac1? a6-eb1b > Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB? 7 > CSeq: 3 REGISTER > PortaBilling: available-funds:7.37 currency:AUD > Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060};? expires=3595 > Server: Sip EXpress router (0.9.6 (i386/freebsd)) > Content-Length: 0 > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Wed Jun 17 00:36:06 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 17:36:06 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617070019.GA5042@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> Message-ID: <20090617073606.GA13623@jdc.jasonjgw.net> Jason White wrote: > The symptom is the following line in outgoing SIP messages while attempting to > establish a call to a gateway via the external profile: > > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org However, if I place an IPv6 call via the internal-ipv6 profile, the o= line contains the correct IPv6 address - so this is only adversely affecting the IPv4 external profile, it seems. From darklion11 at yahoo.com Wed Jun 17 00:44:54 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 17 Jun 2009 00:44:54 -0700 (PDT) Subject: [Freeswitch-users] Allow invites from another sip server? In-Reply-To: <24067552.post@talk.nabble.com> References: <24067552.post@talk.nabble.com> Message-ID: <1245224694870-3091145.post@n2.nabble.com> Thanks a lot... A mistake from me... Edmar Cruz wrote: > > > On my acl.conf.xml > > I allow the ip 116.50.110.2 > > Is this correct? > > > > > > Error sip_invite() ... Error occur rejected by acl domains > > param name="apply-inbound-acl" value="domains"/> > param name="apply-register_acl" value="domains"/> > > -- > View this message in context: > http://www.nabble.com/Allow-invites-from-another-sip-server--tp24067552p24067552.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Allow-invites-from-another-sip-server--tp3090843p3091145.html Sent from the freeswitch-users mailing list archive at Nabble.com. From demuel at thephinix.org Wed Jun 17 01:16:57 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Wed, 17 Jun 2009 09:16:57 +0100 (BST) Subject: [Freeswitch-users] Porta Billing? In-Reply-To: References: Message-ID: <036b0e54c43ce10d1d64bb4db2c2eac9.squirrel@www.thephinix.org> Porta-Billing from PortaSIP? I've tried their stuff and this is when I saw the most disgusting piece of scam software on the VoIP world. Their technical support were the dumbest I ever encountered since time immemorial....Be very beware, they are just extracting your money but their services are shit. > I doubt that header is exposed since its not a standard sip header. > However > you could probably patch mod_sofia to expose it without too much > trouble... > How difficult that would be is dependant on where in session that comes in > > >> From: Mark Campbell-Smith >> Reply-To: >> Date: Wed, 17 Jun 2009 16:58:23 +1000 >> To: >> Subject: [Freeswitch-users] Porta Billing? >> >> Hi! >> >> Does freeswitch support extracting the billing data (PortaBilling) in >> SIP messages? If so, is there anyway I can get that information to an >> extension? >> >> 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis? playMsg: >> Received: >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4? bK1FF90 >> From: {sip:61XXXXXXXXX at sip.pennytel.com};? tag=47E011-580 >> To: {sip:61XXXXXXXXX at sip.pennytel.com};? >> tag=adfde4bc91cd85e752cb0672816ac1? a6-eb1b >> Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB? 7 >> CSeq: 3 REGISTER >> PortaBilling: available-funds:7.37 currency:AUD >> Contact: {sip:61XXXXXXXXX at sip.mydomain.com:5060};? expires=3595 >> Server: Sip EXpress router (0.9.6 (i386/freebsd)) >> Content-Length: 0 >> >> Thanks! >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Wed Jun 17 01:27:16 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 17 Jun 2009 18:27:16 +1000 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: References: <33c87fa30906162358w112779a7n2773533b809175b9@mail.gmail.com> Message-ID: <20090617082716.GA25006@jdc.jasonjgw.net> Ken Rice wrote: > I doubt that header is exposed since its not a standard sip header. However > you could probably patch mod_sofia to expose it without too much trouble... > How difficult that would be is dependant on where in session that comes in Using the info application will reveal whether that header is available in a channel variable. The event mechanism also discloses the channel variables; you can subscribe to events using fs_cli even without writing a script. From jingwei.yang at gmail.com Wed Jun 17 02:39:33 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Wed, 17 Jun 2009 17:39:33 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> Message-ID: <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> Hi Giovanni, Sorry, pretty busy and fully occupied by other stuff today. Have to delay the testing and give you the result tomorrow. Regards, -Jingwei On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang wrote: > Sure, I'll append to you the result tomorrow. > > Regards, > -Jingwei > > > On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli wrote: > >> Hi Jingwei, >> >> Thanks a lot! I'll take care of as soon as possible. >> >> Btw, before I read the Jira, are you testing in linux? >> >> If you are testing on linux, would you please report how it is >> performing under load? I mean, what is the load average with, let say, >> 10 or 20 or more concurrent Skype call? >> >> This has nothing to do with your bug, but will help me in getting >> better performances. >> >> Ciao for now, and thanks again for reporting! >> >> -giovanni >> >> >> >> >> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >> wrote: >> > Hi Giovanni, >> > >> > I've reported it in Jira. Here's the bug url: >> > >> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >> > >> > Thanks, >> > -Jingwei >> > >> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < >> gmaruzz at celliax.org> >> > wrote: >> >> >> >> Hi Jingwel, >> >> thanks for reporting. >> >> >> >> Could you please add a Jira issue with as much details as possible? >> >> >> >> general guide for reporting bugs: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> >> >> what to add for skypiax: >> >> >> >> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >> >> >> >> mod_skypiax Jira: >> >> http://jira.freeswitch.org/browse/MODSKYPIAX >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> ========================================= >> >> www.celliax.org >> >> via Pierlombardo 9, 20135 Milano >> >> Italy >> >> gmaruzz at celliax dot org >> >> Cell : +39-347-2665618 >> >> Fax : +39-02-87390039 >> >> >> >> >> >> >> >> >> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang >> >> wrote: >> >> > Hi Team, >> >> > >> >> > I've been using the record_session feature to record call sessions. >> >> > Here's >> >> > how I prepared the dialplan: >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > And here's how I trigger it: >> >> > >> >> > freeswitch at localhost.localdomain>originate >> skypiax/skypiax2/userAAA >> >> > 2909/userBBB >> >> > >> >> > The call can be established and the data.wav file was generated >> without >> >> > any >> >> > problem. However, once userAAA hung up, a segmentation fault occurred >> >> > and >> >> > freeswitch was automatically shut down. Here are what I got from the >> >> > console: >> >> > >> >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA >> >> > 2909/userBBB >> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >> >> > switch_channel_set_name() >> >> > New Channel skypiax/skypiax2/userAAA >> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >> >> > remote_party_is_ringing() >> >> > Ring-Ready skypiax/skypiax2/userAAA >> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >> >> > outbound_channel_answered() >> >> > Channel [skypiax/skypiax2/userAAA] has been answered >> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >> >> > switch_ivr_session_transfer() >> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >> >> > >> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >> >> > FreeSWITCH->2909/userBBB >> >> > in context default >> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >> >> > switch_channel_set_name() >> >> > New Channel skypiax/ANY/userBBB >> [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >> >> > remote_party_is_ringing() >> >> > Ring-Ready skypiax/ANY/userBBB! >> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >> >> > outbound_channel_answered() >> >> > Channel [skypiax/ANY/userBBB] has been answered >> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >> >> > [CS_EXECUTE] >> >> > [NORMAL_CLEARING] >> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >> Ended >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >> >> > [CS_DESTROY] >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >> >> > [CS_DESTROY] >> >> > Segmentation fault (core dumped) >> >> > >> >> > Please kindly let me know whether there's anything wrong with the >> >> > dialplan >> >> > or the way how I originated the call. >> >> > >> >> > Thanks! >> >> > -Jingwei >> >> > >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/eb3f6ec4/attachment-0002.html From gmaruzz at celliax.org Wed Jun 17 02:58:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 17 Jun 2009 11:58:01 +0200 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> Message-ID: <7b197bef0906170258v1aefcaesd8593737a2e6bc4b@mail.gmail.com> Thanks Jingwei, have a good night! -giovanni On Wed, Jun 17, 2009 at 11:39 AM, Jingwei Yang wrote: > Hi Giovanni, > > Sorry, pretty busy and fully occupied by other stuff today. Have to delay > the testing and give you the result tomorrow. > > Regards, > -Jingwei > > On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang > wrote: >> >> Sure, I'll append to you the result tomorrow. >> >> Regards, >> -Jingwei >> >> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli >> wrote: >>> >>> Hi Jingwei, >>> >>> Thanks a lot! I'll take care of as soon as possible. >>> >>> Btw, before I read the Jira, are you testing in linux? >>> >>> If you are testing on linux, would you please report how it is >>> performing under load? I mean, what is the load average with, let say, >>> 10 or 20 or more concurrent Skype call? >>> >>> This has nothing to do with your bug, but will help me in getting >>> better performances. >>> >>> Ciao for now, and thanks again for reporting! >>> >>> -giovanni >>> >>> >>> >>> >>> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >>> wrote: >>> > Hi Giovanni, >>> > >>> > I've reported it in Jira. Here's the bug url: >>> > >>> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >>> > >>> > Thanks, >>> > -Jingwei >>> > >>> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli >>> > >>> > wrote: >>> >> >>> >> Hi Jingwel, >>> >> thanks for reporting. >>> >> >>> >> Could you please add a Jira issue with as much details as possible? >>> >> >>> >> general guide for reporting bugs: >>> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >> >>> >> what to add for skypiax: >>> >> >>> >> >>> >> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >>> >> >>> >> mod_skypiax Jira: >>> >> http://jira.freeswitch.org/browse/MODSKYPIAX >>> >> >>> >> >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> ========================================= >>> >> www.celliax.org >>> >> via Pierlombardo 9, 20135 Milano >>> >> Italy >>> >> gmaruzz at celliax dot org >>> >> Cell : +39-347-2665618 >>> >> Fax : +39-02-87390039 >>> >> >>> >> >>> >> >>> >> >>> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang >>> >> wrote: >>> >> > Hi Team, >>> >> > >>> >> > I've been using the record_session feature to record call sessions. >>> >> > Here's >>> >> > how I prepared the dialplan: >>> >> > >>> >> > ??? >>> >> > ????? >> >> > expression="^2909/(.*)$"> >>> >> > ??????? >>> >> > ??????? >>> >> > ????? >>> >> > ??? >>> >> > >>> >> > And here's how I trigger it: >>> >> > >>> >> > ??? freeswitch at localhost.localdomain>originate >>> >> > skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > >>> >> > The call can be established and the data.wav file was generated >>> >> > without >>> >> > any >>> >> > problem. However, once userAAA hung up, a segmentation fault >>> >> > occurred >>> >> > and >>> >> > freeswitch was automatically shut down. Here are what I got from the >>> >> > console: >>> >> > >>> >> > freeswitch at localhost.localdomain> originate skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/skypiax2/userAAA >>> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >>> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/skypiax2/userAAA >>> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/skypiax2/userAAA] has been answered >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >>> >> > switch_ivr_session_transfer() >>> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >>> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >>> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >>> >> > >>> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >>> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >>> >> > FreeSWITCH->2909/userBBB >>> >> > in context default >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/ANY/userBBB >>> >> > [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >>> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/ANY/userBBB! >>> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/ANY/userBBB] has been answered >>> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >>> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >>> >> > [CS_EXECUTE] >>> >> > [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >>> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >>> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >>> >> > Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >>> >> > [CS_DESTROY] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >>> >> > [CS_DESTROY] >>> >> > Segmentation fault (core dumped) >>> >> > >>> >> > Please kindly let me know whether there's anything wrong with the >>> >> > dialplan >>> >> > or the way how I originated the call. >>> >> > >>> >> > Thanks! >>> >> > -Jingwei >>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From panselva at gmail.com Wed Jun 17 04:19:03 2009 From: panselva at gmail.com (selva kumar) Date: Wed, 17 Jun 2009 16:49:03 +0530 Subject: [Freeswitch-users] Outboubd is not working while inbound and runs well Message-ID: <45f609f90906170419t74033fcmd878658fd109bdfa@mail.gmail.com> Hi When I add the following line in acl.conf.xml file then inbound works fine. But when I comment these lines only the outbound is working fine.What would be problem? By Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/bb2579f2/attachment-0002.html From dujinfang at gmail.com Wed Jun 17 05:49:05 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 17 Jun 2009 20:49:05 +0800 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24066825.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> Message-ID: comment lines in the user directory do the trick: On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote: > > If FS A has an account 8011105 does FS B also nid to register > 8011105? Yes it > working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option > to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jim at evolutiontel.net Wed Jun 17 06:14:01 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 17 Jun 2009 23:14:01 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue Message-ID: IMHO, as FS is a B2BUA the new leg should state ownership in the SDP. Add to this the fact the IPV6 is blindly copied from leg 1 and the IP address was not decoded correctly there does appear to be a defficiency in the code. - original message - Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue From: Jason White Date: 17/06/2009 07:01 this is an issue which I've been discussing with Brian West on IRC and in e-mail correspondence, which I thought I should bring to the list so that others can look at it as well. The configuration My external SIP profile has its ext-sip-ip and ext-rtp-ip set to stun:stun.freeswitch.org. This is necessary for nat traversal. I have an internal-ipv6 profile as well, which is working, but for some reason it's interfering with calls on the external profile (which of course is an IPv4 profile). The symptom is the following line in outgoing SIP messages while attempting to establish a call to a gateway via the external profile: o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org in response to which, the other side returns a 488 (invalid session description). I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the internal-ipv6.xml profile, but they are said as explained above in the external profile, where the problem lies. This looks like a bug to me but I want to rule out misconfiguration first. This has been tested on revision 13806. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Jun 17 06:16:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:16:04 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617073606.GA13623@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> Message-ID: <132F594D-AF1B-4331-865D-DDD9AEBE53A8@freeswitch.org> Right which is what you have to do... I haven't been able to reproduce the issue... which is odd. /b On Jun 17, 2009, at 2:36 AM, Jason White wrote: > Jason White wrote: > >> The symptom is the following line in outgoing SIP messages while >> attempting to >> establish a call to a gateway via the external profile: >> >> o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org > > However, if I place an IPv6 call via the internal-ipv6 profile, the > o= line > contains the correct IPv6 address - so this is only adversely > affecting the > IPv4 external profile, it seems. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Wed Jun 17 06:17:46 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 17 Jun 2009 15:17:46 +0200 Subject: [Freeswitch-users] UniMRCP - current status? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB939F@cooper> I can see that the UniMRCP libs have been added to FS lately, I was just wondering about the current status/stability for this implementation? And will this be ported to Windows as well? Just curious - since you guys add more features all the time :) /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/dcc71a34/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 17 06:18:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:18:40 -0500 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> Message-ID: <191c3a030906170618v214fb0e9y4d5c64b724a97200@mail.gmail.com> You should not report bugs on the mailing list. please report your bug to jira http://jira.freeswitch.org http://wiki.freeswitch.org/wiki/Reporting_Bugs Make sure you attach a backtrace of your issue and file it under skypeiax so giovanni can track it. On Wed, Jun 17, 2009 at 4:39 AM, Jingwei Yang wrote: > Hi Giovanni, > > Sorry, pretty busy and fully occupied by other stuff today. Have to delay > the testing and give you the result tomorrow. > > Regards, > -Jingwei > > > On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang wrote: > >> Sure, I'll append to you the result tomorrow. >> >> Regards, >> -Jingwei >> >> >> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli > > wrote: >> >>> Hi Jingwei, >>> >>> Thanks a lot! I'll take care of as soon as possible. >>> >>> Btw, before I read the Jira, are you testing in linux? >>> >>> If you are testing on linux, would you please report how it is >>> performing under load? I mean, what is the load average with, let say, >>> 10 or 20 or more concurrent Skype call? >>> >>> This has nothing to do with your bug, but will help me in getting >>> better performances. >>> >>> Ciao for now, and thanks again for reporting! >>> >>> -giovanni >>> >>> >>> >>> >>> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >>> wrote: >>> > Hi Giovanni, >>> > >>> > I've reported it in Jira. Here's the bug url: >>> > >>> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >>> > >>> > Thanks, >>> > -Jingwei >>> > >>> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < >>> gmaruzz at celliax.org> >>> > wrote: >>> >> >>> >> Hi Jingwel, >>> >> thanks for reporting. >>> >> >>> >> Could you please add a Jira issue with as much details as possible? >>> >> >>> >> general guide for reporting bugs: >>> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >> >>> >> what to add for skypiax: >>> >> >>> >> >>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >>> >> >>> >> mod_skypiax Jira: >>> >> http://jira.freeswitch.org/browse/MODSKYPIAX >>> >> >>> >> >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> ========================================= >>> >> www.celliax.org >>> >> via Pierlombardo 9, 20135 Milano >>> >> Italy >>> >> gmaruzz at celliax dot org >>> >> Cell : +39-347-2665618 >>> >> Fax : +39-02-87390039 >>> >> >>> >> >>> >> >>> >> >>> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang>> > >>> >> wrote: >>> >> > Hi Team, >>> >> > >>> >> > I've been using the record_session feature to record call sessions. >>> >> > Here's >>> >> > how I prepared the dialplan: >>> >> > >>> >> > >>> >> > >> expression="^2909/(.*)$"> >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > And here's how I trigger it: >>> >> > >>> >> > freeswitch at localhost.localdomain>originate >>> skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > >>> >> > The call can be established and the data.wav file was generated >>> without >>> >> > any >>> >> > problem. However, once userAAA hung up, a segmentation fault >>> occurred >>> >> > and >>> >> > freeswitch was automatically shut down. Here are what I got from the >>> >> > console: >>> >> > >>> >> > freeswitch at localhost.localdomain> originate >>> skypiax/skypiax2/userAAA >>> >> > 2909/userBBB >>> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/skypiax2/userAAA >>> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >>> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/skypiax2/userAAA >>> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/skypiax2/userAAA] has been answered >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >>> >> > switch_ivr_session_transfer() >>> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >>> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >>> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >>> >> > >>> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >>> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >>> >> > FreeSWITCH->2909/userBBB >>> >> > in context default >>> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >>> >> > switch_channel_set_name() >>> >> > New Channel skypiax/ANY/userBBB >>> [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >>> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >>> >> > remote_party_is_ringing() >>> >> > Ring-Ready skypiax/ANY/userBBB! >>> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >>> >> > outbound_channel_answered() >>> >> > Channel [skypiax/ANY/userBBB] has been answered >>> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >>> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >>> >> > [CS_EXECUTE] >>> >> > [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >>> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >>> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >>> Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >>> >> > [CS_DESTROY] >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >>> >> > [CS_DESTROY] >>> >> > Segmentation fault (core dumped) >>> >> > >>> >> > Please kindly let me know whether there's anything wrong with the >>> >> > dialplan >>> >> > or the way how I originated the call. >>> >> > >>> >> > Thanks! >>> >> > -Jingwei >>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> > >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/db5354a2/attachment-0002.html From brian at freeswitch.org Wed Jun 17 06:24:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:24:02 -0500 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: References: Message-ID: While the header looks valid it should be an X-Header then it would show up. /b On Jun 17, 2009, at 2:32 AM, Ken Rice wrote: > I doubt that header is exposed since its not a standard sip header. > However > you could probably patch mod_sofia to expose it without too much > trouble... > How difficult that would be is dependant on where in session that > comes in From brian at freeswitch.org Wed Jun 17 06:25:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:25:09 -0500 Subject: [Freeswitch-users] UniMRCP - current status? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB939F@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB939F@cooper> Message-ID: <9584E028-F7D9-4020-9997-2E8680C083BC@freeswitch.org> No, we haven't done the solution file for the module on windows but the lib has been done on windows. It still need more testing and such but its functional. /b On Jun 17, 2009, at 8:17 AM, Peter Olsson wrote: > I can see that the UniMRCP libs have been added to FS lately, I was > just wondering about the current status/stability for this > implementation? And will this be ported to Windows as well? > > Just curious ? since you guys add more features all the time :) > > /Peter > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/1ca4439c/attachment-0002.html From brian at freeswitch.org Wed Jun 17 06:26:13 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:26:13 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: References: Message-ID: <931BB0A4-DF71-4E3C-9968-8AED09C46DFF@freeswitch.org> On Jun 17, 2009, at 8:14 AM, Jim Burke wrote: > IMHO, as FS is a B2BUA the new leg should state ownership in the > SDP. Add to this the fact the IPV6 is blindly copied from leg 1 and > the IP address was not decoded correctly there does appear to be a > defficiency in the code. I don't think that is what is going on unless you're trying to do proxy media from IPv4 to IPv6 which I haven't ever tried nor do I recommend. /b From anthony.minessale at gmail.com Wed Jun 17 06:27:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:27:03 -0500 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: References: Message-ID: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> 1) look in the wiki and try to understand the difference between application and FSAPI functions. 2) use apiExecute function from js to execute FSAPI functions such as uuid_kill On Wed, Jun 17, 2009 at 1:44 AM, Baskar wrote: > *Hi, > > I have some questions can any one assist me and answer this query > > Question 1: Can we able to execute all the api command through JavaScript > using session.execute. > > Question 2: How to kill the session using uuid_kill whether it is > possible. **If yes means how we will use uuid_kill in session* > * > i have tried some thing like this : > > session.execute("uuid_kill", "session.uuid"); > > session.execute("uuid_kill session.uuid"); > > but there is no output for the session.execute. > > Whether it is possible to execute uuid_kill in javascript > > Thanks in advance > > > Warm Regards, > N.Baskar > > * > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/bf13cb23/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 17 06:32:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:32:42 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617070019.GA5042@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> Message-ID: <191c3a030906170632s39130c29m717324660f0d0fbb@mail.gmail.com> can you confirm that they are not being inherited by some of the helper variables exported from vars.xml? can you minimalise your ip4 profile by cutting all the commented lines and actually remove the ext-rtp-ip and ext-sip-ip params? Then possibly post your config for each profile. On Wed, Jun 17, 2009 at 2:00 AM, Jason White wrote: > this is an issue which I've been discussing with Brian West on IRC and in > e-mail correspondence, which I thought I should bring to the list so that > others can look at it as well. > > The configuration > > My external SIP profile has its ext-sip-ip and ext-rtp-ip set to > stun:stun.freeswitch.org. This is necessary for nat traversal. > > I have an internal-ipv6 profile as well, which is working, but for some > reason > it's interfering with calls on the external profile (which of course is an > IPv4 profile). > > The symptom is the following line in outgoing SIP messages while attempting > to > establish a call to a gateway via the external profile: > > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org > > in response to which, the other side returns a 488 (invalid session > description). > > I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the > internal-ipv6.xml profile, but they are said as explained above in the > external profile, where the problem lies. > > This looks like a bug to me but I want to rule out misconfiguration first. > > This has been tested on revision 13806. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/abb0e0c2/attachment-0002.html From brian at freeswitch.org Wed Jun 17 06:36:22 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:36:22 -0500 Subject: [Freeswitch-users] outbound error log In-Reply-To: <45f609f90906162314q3b450f4aid4d98c81af9a1e8f@mail.gmail.com> References: <45f609f90906162314q3b450f4aid4d98c81af9a1e8f@mail.gmail.com> Message-ID: The link would be helpful. /b On Jun 17, 2009, at 1:14 AM, selva kumar wrote: > > Hi Michael, > I have pasted the freeswitch logs as requested in > (pastebin.freeswitch.org) > > > Thanks > Sam > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/aee53dee/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 17 06:36:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 17 Jun 2009 08:36:35 -0500 Subject: [Freeswitch-users] Force SIP UA to pick up call during ringing? In-Reply-To: <4A37FC59.60401@gmx.net> References: <4A36E52D.2010005@gmx.net> <4A3764BB.3040403@gmx.net> <4A37923D.8060809@gmx.net> <66C813CD-CC79-4A72-91ED-04BC14132851@jerris.com> <4A37DB1C.8070706@gmx.net> <3AB2D06C-145E-408D-AD20-2E4D7BD4D553@jerris.com> <4A37FC59.60401@gmx.net> Message-ID: <191c3a030906170636s275378bey40ce3eca6b5d99a4@mail.gmail.com> Looks like you are trying to build a call center have you seen mod_fifo? It's designed to let people on headsets sit idle and you can send calls to them at will. On Tue, Jun 16, 2009 at 3:11 PM, Peter P GMX wrote: > Thanks Michael, > > I have disabled it now. > > I finally got it to work, (sip_h_Call-Info=;answer-after=0) > but the behaviour was not as desired, as I didn't manage the phone to > pick up the call on the headset. It will only have the speaker enabled. > So I will have to go a different way with parking the call and then > forward it. > > Best regards > Peter > > > Michael Jerris schrieb: > > uuid_setvar sip_invite_params intercom=true should be > > unnecessary. > > > > Mike > > > > On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote: > > > > > >> It mainly works now by uuid_transfer the following way via event > >> socket. > >> uuid_setvar sip_invite_params intercom=true > >> uuid_setvar sip_auto_answer true > >> uuid_transfer 1000 XML default > >> so the call is transferred from 1000 to 1000. > >> > >> What happens: > >> 1) If I disable intercom on the Snom phone, the phone rings, stops > >> ringing and rings again (ok) > >> 1) If I enable intercom on the Snom phone, the phone rings, stops > >> ringing and hangs up (not ok) > >> > >> So I do not get the Snom to pick up the call in intercom mode. > >> > >> The last invite is: > >> INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib > >> SIP/2.0 > >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > >> Route: ;transport=tls;line=er6kxnib > >> Max-Forwards: 68 > >> From: "Peter FS" ;tag=9eQ8rjQy533HF > >> To: > >> >> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > >> CSeq: 116467629 INVITE > >> Contact: > >> Call-Info: ;answer-after=0 > >> The intercom part is there and the Call-Info line with answer-after > >> also. > >> > >> The phone answers with > >> SIP/2.0 401 Unauthorized > >> Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF > >> From: "Peter FS" ;tag=9eQ8rjQy533HF > >> To: > >> >> > >>> ;tag=71rskygkr2 > >>> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > >> CSeq: 116467629 INVITE > >> Contact: > >> ;reg-id=1 > >> WWW-Authenticate: Digest realm="sip2.mycompany.de", > >> nonce="2ee26efe6ab27f88", algorithm=MD5 > >> Content-Length: 0 > >> and hangs up. > >> > >> Anybody know how to solve this Snom intercom issue? > >> > >> Best regards > >> Peter > >> > >> > >> Michael Jerris schrieb: > >> > >>> The transfer should work but it sounds like offhook agents is what > >>> your really trying to accomplish so I would go with brian's > >>> suggestion. > >>> > >>> > >>> > >>> On Jun 16, 2009, at 7:38 AM, Peter P GMX > >>> wrote: > >>> > >>> > >>> > >>>> Hello Michael, > >>>> > >>>> I want the phone be ringing, just for acoustical feedback reasons. > >>>> > >>>> But what if I > >>>> > >>>> * transfer it to the same user destination again (now with > >>>> intercom > >>>> enabled), will this work? > >>>> * transfer it to park and then transfer it to the same destination > >>>> again (now with intercom enabled) > >>>> > >>>> Best regards > >>>> Peter > >>>> > >>>> Michael Jerris schrieb: > >>>> > >>>> > >>>>> The only way I can think to do this today would be to cancel the > >>>>> call > >>>>> and re send with the intercom headers for a phone that supports it. > >>>>> It may be possible to send a reinvite with autoanswer headers but I > >>>>> doubt that would work, all you could do is try making code to do it > >>>>> it > >>>>> a sipp or sipsak scenario and test it. A better aproach might be > >>>>> to > >>>>> answer the call normally and detect that to start your web workflow > >>>>> or > >>>>> not really ring the phone, just the web app and deliver the call > >>>>> with > >>>>> autoanswer when the button is hit in the web ui. > >>>>> > >>>>> Mike > >>>>> > >>>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX > >>>>> wrote: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>> Hello Brian, > >>>>>> > >>>>>> this is too easy :-). > >>>>>> > >>>>>> This is for a small callcenter app and I only want the user to > >>>>>> pickup > >>>>>> the call once (to accept the call in X-Lite (or a Snom phone) > >>>>>> and to > >>>>>> start the workflow on the web application). I do not want him to > >>>>>> accept > >>>>>> the call on the phone and then on the Web app. > >>>>>> > >>>>>> Best regards > >>>>>> Peter > >>>>>> > >>>>>> > >>>>>> > >>>>>> Brian West schrieb: > >>>>>> > >>>>>> > >>>>>> > >>>>>>> click on the AA button? :) > >>>>>>> > >>>>>>> /b > >>>>>>> > >>>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>> What is the best way to have this done? Move the call to park > >>>>>>>> and > >>>>>>>> then > >>>>>>>> retransfer again with intercom, or is there a better solution? > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>> _______________________________________________ > >>>>>>> Freeswitch-users mailing list > >>>>>>> Freeswitch-users at lists.freeswitch.org > >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>>> http://www.freeswitch.org > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> _______________________________________________ > >>>>>> Freeswitch-users mailing list > >>>>>> Freeswitch-users at lists.freeswitch.org > >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> http://www.freeswitch.org > >>>>>> > >>>>>> > >>>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/6f9ffc12/attachment-0002.html From rdenert at tng.de Wed Jun 17 06:42:41 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 17 Jun 2009 15:42:41 +0200 (CEST) Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <15918673.24821245245881356.JavaMail.root@zimbra.tng.de> Message-ID: <12807519.24911245246161312.JavaMail.root@zimbra.tng.de> Hello, I have one question about including a webserver. I made a dialplan which I deposited on a webserver. That was no the problem. But I have a commandline in the dialplan that doesn't work correct. The fs_cli says: (this line has no problem. I just mentioned this for the sake of completeness) My features.xml looks like this: In my freeswitch server this line works perfekt. The extension "test" in features.xml is executed. For near information: When I call the FS (it is a conferencing system) the machine looks in the dialplan which is deposited on a webserver. Than I press the digits into phone to get access in the conferencing room. But than I get two important messages in the fs_cli: 2009-06-17 15:38:21 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 123456789->test in context features 2009-06-17 15:38:21 [WARNING] mod_dialplan_xml.c:263 dialplan_hunt() Context features not found Does anybody has an idea? Greetz -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From brian at freeswitch.org Wed Jun 17 06:47:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 08:47:57 -0500 Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <12807519.24911245246161312.JavaMail.root@zimbra.tng.de> References: <12807519.24911245246161312.JavaMail.root@zimbra.tng.de> Message-ID: <78F65C7F-F4E6-4CB4-ADEE-3557385E8039@freeswitch.org> Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > Context features not found From timb0311 at hotmail.com Wed Jun 17 07:00:25 2009 From: timb0311 at hotmail.com (Tim B) Date: Wed, 17 Jun 2009 10:00:25 -0400 Subject: [Freeswitch-users] Mod_Fax / TxFax / Originate In-Reply-To: References: Message-ID: Trying to do a local test for faxing. Keep getting an error. default dialplan: //inbound from remote box works fine - connect asterisk box and fs box, then fax from asterisk to fs... OK - also fax from fs to asterisk.... OK // local fax on fs .... FAILS!! // my originate command: originate sofia/internal/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) // error as follows: 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing FreeSWITCH->8000 in context public 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 Legged calls 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] _________________________________________________________________ Lauren found her dream laptop. Find the PC that?s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/81096a52/attachment-0002.html From krice at suspicious.org Wed Jun 17 08:15:44 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 17 Jun 2009 10:15:44 -0500 Subject: [Freeswitch-users] Porta Billing? In-Reply-To: <20090617082716.GA25006@jdc.jasonjgw.net> Message-ID: While you are correct Jason, this particular header is not a valid SIP Header... It should be prefixed with X- ie: "X-Some-Custom-Header: foo" When you have non-standard headers lib sofia will stick them in a struct on the backend for us to manually deal with but they are not auto-magically assigned to channel variables unless they are X- and P- headers (P- headers are some special meaning headers typically associated with billing and privacy or are "private" headers > From: Jason White > > Using the info application will reveal whether that header is available in a > channel variable. > > The event mechanism also discloses the channel variables; you can subscribe to > events using fs_cli even without writing a script. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonny.voip at gmail.com Wed Jun 17 08:19:35 2009 From: jonny.voip at gmail.com (Jonathan DiVita) Date: Wed, 17 Jun 2009 11:19:35 -0400 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita) In-Reply-To: References: Message-ID: I was able to compile mod_opal with the patches you suggested. Now, once that is done is there anything more I need to do inorder for mod_opal to show up when I run freeswitch? I didn't see that it had started when I ran freeswitch after the mod_opal compile. Thanks! Jon ----- Original Message ----- From: To: Sent: Wednesday, June 17, 2009 12:26 AM Subject: Freeswitch-users Digest, Vol 36, Issue 159 > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Compiling Issues: Opal with Latest SVN Builds 6-19-09 > (Brian West) > 2. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 3. Re: session.getDigits() not working (Brian West) > 4. Re: How can I join two freeswitch on two servers? (Brian West) > 5. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 6. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 7. Re: How can I join two freeswitch on two servers? (Brian West) > 8. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 16 Jun 2009 22:12:01 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN > Builds 6-19-09 > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="iso-8859-1" > > please see MODOPAL-10 on jira. > > /b > > On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > >> Hello, all. I'm currently playing around with a new install of >> Freeswitch and wanted to try out mod_opal. Below are the current >> SVN builds for opal, ptlib, and freeswitch. I end up with the >> following errors when compiling. >> >> making all mod_opal >> Compiling mod_opal.cpp... >> Compiling mod_opal.cpp ... >> In file included from mod_opal.cpp:25: >> mod_opal.h:151: error: conflicting return type specified for >> ?virtual OpalLocalConnection* >> FSEndPoint::CreateConnection(OpalCall&, void*)? >> /usr/include/opal/opal/localep.h:267: error: overriding ?virtual >> ptlib_virtual_function_changed_or_removed****** >> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? >> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, >> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, >> switch_channel_t*)?: >> mod_opal.cpp:564: error: no matching function for call to >> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, >> NULL)? >> /usr/include/opal/opal/localep.h:290: note: candidates are: >> OpalLocalConnection::OpalLocalConnection(OpalCall&, >> OpalLocalEndPoint&, void*, unsigned int, >> OpalConnection::StringOptions*, char) >> /usr/include/opal/opal/localep.h:276: note: >> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) >> make[4]: *** [mod_opal.lo] Error 1 >> make[3]: *** [all] Error 1 >> make[2]: *** [mod_opal-all] Error 1 >> make[1]: *** [mod_opal] Error 2 >> make: *** [mod_opal] Error 2 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <929908.28142.qm at web57310.mail.re1.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > How can i turn off authentication? This is my acl.conf.xml on > 192.168.0.105 > > > > > > > On 192.168.0.4 > > > > > > > ________________________________ > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, June 16, 2009 10:49:58 PM > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > > Turn off authentication or use ACL's > > /b > > On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > >> Is there another way to manage the gateway with the caller id of the >> user >> not the gateway user id and is there a gateway that doesn't need a >> username >> and password? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/2839a86d/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Tue, 16 Jun 2009 22:31:21 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] session.getDigits() not working > To: freeswitch-users at lists.freeswitch.org > Message-ID: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Shouldn't have really changed any behavior at all... What svn rev are > you on? > > /b > > On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > >> API CALL [global_getvar()] output: >> external_ssl_enable=false >> external_tls_port=5081 >> external_sip_port=5080 >> external_auth_calls=false >> internal_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_sip_port=5060 >> default_provider_contact=5000 >> default_provider_from_domain=example.com >> default_provider_password=password >> external_rtp_ip=74.92.196.241 >> xmpp_server_profile=xmpps >> xmpp_client_profile=xmppc >> global_codec_prefs=G722,PCMU,PCMA,GSM >> hold_music=local_stream://moh >> external_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_auth_calls=true >> local_ip_v4=192.168.0.40 >> unroll_loops=true >> default_areacode=918 >> default_provider_register=false >> local_mask_v4=255.255.255.0 >> default_password=1234 >> call_debug=false >> local_ip_v6=::1 >> default_provider_username=joeuser >> sound_prefix=/var/opt/freeswitch/sounds/en/us/callie >> outbound_caller_id=0000000000 >> default_country=US >> base_dir=/var/opt/freeswitch >> bind_server_ip=auto >> internal_tls_port=5061 >> switch_serial=c0a8002854db >> default_provider=example.com >> outbound_codec_prefs=PCMU,PCMA,GSM >> domain_name=192.168.0.40 >> domain=192.168.0.40 >> external_sip_ip=74.92.196.241 >> outbound_caller_name=Versafon.com >> rs-ring=%(1000, 4000, 425.0, 0.0) >> sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) >> internal_ssl_enable=false >> console_loglevel=debug >> uk-ring=%(400,200,400,450);%(400,2200,400,450) >> us-ring=%(2000, 4000, 440.0, 480.0) >> sip_tls_version=tlsv1 >> fr-ring=%(1500, 3500, 440.0, 0.0) >> bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) >> > > > > > ------------------------------ > > Message: 4 > Date: Tue, 16 Jun 2009 22:36:38 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/47e21916/attachment-0001.html > > ------------------------------ > > Message: 5 > Date: Tue, 16 Jun 2009 21:00:04 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 6 > Date: Tue, 16 Jun 2009 21:02:21 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > Like this? > > I put this on external profile > > "/> > "/> > > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 7 > Date: Tue, 16 Jun 2009 23:08:46 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <93456DA1-47C0-4524-903B-0FDE310EE93D at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > COPY paste fail :) > > > > something like that as per the example. > > /b > > On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > >> >> How can sofia profile can call ACL? >> Can you give me an example? >> Like this? >> >> I put this on external profile >> >> "/> >> "/> >> >> >> Brian West-3 wrote: >>> >>> Now you have to tell the sofia profile to use that ACL >>> >>> /b >>> >>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>> >>>> How can i turn off authentication? This is my acl.conf.xml on >>>> 192.168.0.105 >>>> >>>> >>>> >>>> >>>> >>>> On 192.168.0.4 >>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > ------------------------------ > > Message: 8 > Date: Tue, 16 Jun 2009 21:26:38 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066825.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > If FS A has an account 8011105 does FS B also nid to register 8011105? Yes > it > working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option > to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 159 > ************************************************* From Claudio.Cavalera at italtel.it Wed Jun 17 08:20:19 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 17 Jun 2009 17:20:19 +0200 Subject: [Freeswitch-users] Freeswitch as a B2B Application Server for IMS Message-ID: Hello freeswitchers, lately I'm trying to set up a testbed to ivestigate a potential use of freeswitch as a Back-to-Back application server in an IMS architecture. I've seen IMS specs are also linked here http://wiki.freeswitch.org/wiki/Documentation so I've thinked maybe there's a chance :-) I also have the inner feeling that fs could do an amazing job in IMS field as he does in NGN. For my testing I'm now using OpenIMSCore as control layer (where phones do register) and I'm trying to put fs on top of it as a b2b application server to provide services. I would like to share with you my experience to see if something could be done about this scenario (or if it's worth the trouble at least). Alice and Bob are two users registered to the IMS Core and they both have a profile for which originating and terminating INVITEs get triggered towards the fs application server. When Alice calls Bob the call setup would include three legs: 1) Alice -> PCSCF (orig) -> SCSCF (orig) -> FS (orig) 2) FS (orig) -> SCSCF (orig) -> ICSCF -> SCSCF (term) -> FS (term) 3) FS (term) -> SCSCF (term) -> PCSCF(term) -> Bob This partially works already with fs out of the box, but there are a still a few issues to be solved. When FS starts the brand new leg 2) as a B2B User Agent he should keep the Route: header in the SIP INVITE "almost the same" as the one he received in the leg 1) I see here two different issues: a) Getting the Route: header out of incoming invite in leg 1) b) Writing the proper Route: header and have FS behaving correctly at transport layer in the outgoing INVITE in leg 2) a) Now please correct me if I'm wrong: at the moment the header is not a channel variable available in fs (e.g. I don't get it with the "info" app). It there were a way to get this header out of the incoming INVITEs, I could do the logic to parse it and forge a proper one in the outgoing INVITE. b) Concerning how to write the header, I'm already working with fs_path directive which also makes FS behaves correctly at network layer. Could someone please elaborate a little bit about the alternative to fs_path directive? I've seen there are already many in theory: - combining sip_h_Route= with http://wiki.freeswitch.org/wiki/Variable_sip_network_destination - use of http://wiki.freeswitch.org/wiki/Variable_sip_route_uri - use of fs_path= http://wiki.freeswitch.org/wiki/Sofia#Specifying_SIP_Proxy_With_fs_path I've simplified the scenario a little bit, there are other things that the B2B AS should do (e.g. removing Record-Route:) but FS do them already from what I've tested. If anyone in the community is interested I'm here to provide further information or share my experience to implement the best solution. Best regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From peter.olsson at visionutveckling.se Wed Jun 17 08:29:44 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 17 Jun 2009 17:29:44 +0200 Subject: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita) In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB93DF@cooper> If you use latest stable opal package you will be able to compile without patching mod_opal. That is what the guys from Opal recommend. The latest trunk has lots of things going on, so I'm not sure the patches I have supplied are sufficient anymore.. To load the module you must also enable it as a module in modules.conf.xml. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jonathan DiVita Skickat: den 17 juni 2009 17:20 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita) I was able to compile mod_opal with the patches you suggested. Now, once that is done is there anything more I need to do inorder for mod_opal to show up when I run freeswitch? I didn't see that it had started when I ran freeswitch after the mod_opal compile. Thanks! Jon ----- Original Message ----- From: To: Sent: Wednesday, June 17, 2009 12:26 AM Subject: Freeswitch-users Digest, Vol 36, Issue 159 > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Compiling Issues: Opal with Latest SVN Builds 6-19-09 > (Brian West) > 2. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 3. Re: session.getDigits() not working (Brian West) > 4. Re: How can I join two freeswitch on two servers? (Brian West) > 5. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 6. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 7. Re: How can I join two freeswitch on two servers? (Brian West) > 8. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 16 Jun 2009 22:12:01 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN > Builds 6-19-09 > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="iso-8859-1" > > please see MODOPAL-10 on jira. > > /b > > On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > >> Hello, all. I'm currently playing around with a new install of >> Freeswitch and wanted to try out mod_opal. Below are the current >> SVN builds for opal, ptlib, and freeswitch. I end up with the >> following errors when compiling. >> >> making all mod_opal >> Compiling mod_opal.cpp... >> Compiling mod_opal.cpp ... >> In file included from mod_opal.cpp:25: >> mod_opal.h:151: error: conflicting return type specified for >> ?virtual OpalLocalConnection* >> FSEndPoint::CreateConnection(OpalCall&, void*)? >> /usr/include/opal/opal/localep.h:267: error: overriding ?virtual >> ptlib_virtual_function_changed_or_removed****** >> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? >> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, >> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, >> switch_channel_t*)?: >> mod_opal.cpp:564: error: no matching function for call to >> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, >> NULL)? >> /usr/include/opal/opal/localep.h:290: note: candidates are: >> OpalLocalConnection::OpalLocalConnection(OpalCall&, >> OpalLocalEndPoint&, void*, unsigned int, >> OpalConnection::StringOptions*, char) >> /usr/include/opal/opal/localep.h:276: note: >> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) >> make[4]: *** [mod_opal.lo] Error 1 >> make[3]: *** [all] Error 1 >> make[2]: *** [mod_opal-all] Error 1 >> make[1]: *** [mod_opal] Error 2 >> make: *** [mod_opal] Error 2 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/a7bb947f/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Tue, 16 Jun 2009 20:03:13 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <929908.28142.qm at web57310.mail.re1.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > How can i turn off authentication? This is my acl.conf.xml on > 192.168.0.105 > > > > > > > On 192.168.0.4 > > > > > > > ________________________________ > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, June 16, 2009 10:49:58 PM > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > > Turn off authentication or use ACL's > > /b > > On Jun 16, 2009, at 8:28 PM, Edmar Cruz wrote: > >> Is there another way to manage the gateway with the caller id of the >> user >> not the gateway user id and is there a gateway that doesn't need a >> username >> and password? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/2839a86d/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Tue, 16 Jun 2009 22:31:21 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] session.getDigits() not working > To: freeswitch-users at lists.freeswitch.org > Message-ID: <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Shouldn't have really changed any behavior at all... What svn rev are > you on? > > /b > > On Jun 16, 2009, at 5:50 PM, paul.degt wrote: > >> API CALL [global_getvar()] output: >> external_ssl_enable=false >> external_tls_port=5081 >> external_sip_port=5080 >> external_auth_calls=false >> internal_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_sip_port=5060 >> default_provider_contact=5000 >> default_provider_from_domain=example.com >> default_provider_password=password >> external_rtp_ip=74.92.196.241 >> xmpp_server_profile=xmpps >> xmpp_client_profile=xmppc >> global_codec_prefs=G722,PCMU,PCMA,GSM >> hold_music=local_stream://moh >> external_ssl_dir=/var/opt/freeswitch/conf/ssl >> internal_auth_calls=true >> local_ip_v4=192.168.0.40 >> unroll_loops=true >> default_areacode=918 >> default_provider_register=false >> local_mask_v4=255.255.255.0 >> default_password=1234 >> call_debug=false >> local_ip_v6=::1 >> default_provider_username=joeuser >> sound_prefix=/var/opt/freeswitch/sounds/en/us/callie >> outbound_caller_id=0000000000 >> default_country=US >> base_dir=/var/opt/freeswitch >> bind_server_ip=auto >> internal_tls_port=5061 >> switch_serial=c0a8002854db >> default_provider=example.com >> outbound_codec_prefs=PCMU,PCMA,GSM >> domain_name=192.168.0.40 >> domain=192.168.0.40 >> external_sip_ip=74.92.196.241 >> outbound_caller_name=Versafon.com >> rs-ring=%(1000, 4000, 425.0, 0.0) >> sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) >> internal_ssl_enable=false >> console_loglevel=debug >> uk-ring=%(400,200,400,450);%(400,2200,400,450) >> us-ring=%(2000, 4000, 440.0, 480.0) >> sip_tls_version=tlsv1 >> fr-ring=%(1500, 3500, 440.0, 0.0) >> bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440) >> > > > > > ------------------------------ > > Message: 4 > Date: Tue, 16 Jun 2009 22:36:38 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Now you have to tell the sofia profile to use that ACL > > /b > > On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: > >> How can i turn off authentication? This is my acl.conf.xml on >> 192.168.0.105 >> >> >> >> >> >> On 192.168.0.4 >> >> >> >> > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090616/47e21916/attachment-0001.html > > ------------------------------ > > Message: 5 > Date: Tue, 16 Jun 2009 21:00:04 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 6 > Date: Tue, 16 Jun 2009 21:02:21 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066647.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > How can sofia profile can call ACL? > Can you give me an example? > Like this? > > I put this on external profile > > "/> > "/> > > > Brian West-3 wrote: >> >> Now you have to tell the sofia profile to use that ACL >> >> /b >> >> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >> >>> How can i turn off authentication? This is my acl.conf.xml on >>> 192.168.0.105 >>> >>> >>> >>> >>> >>> On 192.168.0.4 >>> >>> >>> >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 7 > Date: Tue, 16 Jun 2009 23:08:46 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <93456DA1-47C0-4524-903B-0FDE310EE93D at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > COPY paste fail :) > > > > something like that as per the example. > > /b > > On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: > >> >> How can sofia profile can call ACL? >> Can you give me an example? >> Like this? >> >> I put this on external profile >> >> "/> >> "/> >> >> >> Brian West-3 wrote: >>> >>> Now you have to tell the sofia profile to use that ACL >>> >>> /b >>> >>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>> >>>> How can i turn off authentication? This is my acl.conf.xml on >>>> 192.168.0.105 >>>> >>>> >>>> >>>> >>>> >>>> On 192.168.0.4 >>>> >>>> >>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > ------------------------------ > > Message: 8 > Date: Tue, 16 Jun 2009 21:26:38 -0700 (PDT) > From: Edmar Cruz > Subject: Re: [Freeswitch-users] How can I join two freeswitch on two > servers? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <24066825.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > If FS A has an account 8011105 does FS B also nid to register 8011105? Yes > it > working on a gateway but the username of the gateway was shown on my > softphone and also it nids a password for the gateway... is there an > option > to view the caller name and number of the FS A gateway to FS B? > > > > > Brian West-3 wrote: >> >> COPY paste fail :) >> >> >> >> something like that as per the example. >> >> /b >> >> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >> >>> >>> How can sofia profile can call ACL? >>> Can you give me an example? >>> Like this? >>> >>> I put this on external profile >>> >>> "/> >>> "/> >>> >>> >>> Brian West-3 wrote: >>>> >>>> Now you have to tell the sofia profile to use that ACL >>>> >>>> /b >>>> >>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>> >>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>> 192.168.0.105 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On 192.168.0.4 >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 159 > ************************************************* _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a390b1b32934139097269! From paul.degt at gmail.com Wed Jun 17 08:52:31 2009 From: paul.degt at gmail.com (paul.degt) Date: Wed, 17 Jun 2009 11:52:31 -0400 Subject: [Freeswitch-users] session.getDigits() not working In-Reply-To: References: <4A35C72A.6030804@gmail.com> <4A378D90.2040209@gmail.com> <4A3821CB.2070904@gmail.com> <84E4DE1F-D3B5-48B6-B420-AFB301CE1F9C@freeswitch.org> <4A387874.5090105@gmail.com> Message-ID: <4A39113F.8080702@gmail.com> I will in a couple of days and report back. Brian West wrote: > can you update and try that again? > > /b > > On Jun 17, 2009, at 12:00 AM, paul.degt at gmail.com wrote: > > >> 13564 >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From larclap at yahoo.com Wed Jun 17 10:06:50 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 17 Jun 2009 10:06:50 -0700 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? Message-ID: <012901c9ef6e$01b3ee60$051bcb20$@com> In conf/dialplan/default.xml, the eavesdrop extension's condition is - expression="^88(.*)$|^\*0(.*)$"> Is this intended? I thought it was defined to eavesdrop on internal extensions. Why wouldn't it be something like - expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop extension. Log: 1030 Dialplan: sofia/internal/1009 at 192.168.10.29 parsing [default->eavesdrop] continue=false 1031 Dialplan: sofia/internal/1009 at 192.168.10.29 Regex (PASS) [eavesdrop] destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false 1032 Dialplan: sofia/internal/1009 at 192.168.10.29 Action answer() Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/947d075d/attachment-0002.html From msc at freeswitch.org Wed Jun 17 10:22:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Jun 2009 12:22:46 -0500 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? In-Reply-To: <012901c9ef6e$01b3ee60$051bcb20$@com> References: <012901c9ef6e$01b3ee60$051bcb20$@com> Message-ID: <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb wrote: > In conf/dialplan/default.xml, the eavesdrop extension's condition is - > expression="^88(.*)$|^\*0(.*)$"> > > > > Is this intended? I thought it was defined to eavesdrop on internal > extensions. Why wouldn't it be something like - > expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial > 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop > extension. > > > Lars, "8885551212" should never match regex expression "^88(\d{3,5})$" . It looks to me like your eavesdrop is matching on "^88(.*)$" which most definitely WILL match 8885551212. Please check your dialplan's eavesdrop extension's regex and make sure it is correct. -MC > Log: > > 1030 Dialplan: sofia/internal/1009 at 192..168.10.29parsing [default->eavesdrop] continue=false > > 1031 Dialplan: sofia/internal/1009 at 192..168.10.29Regex (PASS) [eavesdrop] destination_number(8885819795) =~ > /^88(.*)$|^\*0(.*)$/ break=on-false > > 1032 Dialplan: sofia/internal/1009 at 192..168.10.29Action answer() > > > > Thanks, Lars > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/d982db85/attachment-0002.html From diego.viola at gmail.com Wed Jun 17 10:26:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 17 Jun 2009 13:26:20 -0400 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> References: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> Message-ID: <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> Applications are the ones in mod_dptools and FSAPI are mod_commands API right? On Wed, Jun 17, 2009 at 9:27 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > 1) look in the wiki and try to understand the difference between > application and FSAPI functions. > 2) use apiExecute function from js to execute FSAPI functions such as > uuid_kill > > > > On Wed, Jun 17, 2009 at 1:44 AM, Baskar wrote: > >> *Hi, >> >> I have some questions can any one assist me and answer this query >> >> Question 1: Can we able to execute all the api command through JavaScript >> using session.execute. >> >> Question 2: How to kill the session using uuid_kill whether it is >> possible. **If yes means how we will use uuid_kill in session* >> * >> i have tried some thing like this : >> >> session.execute("uuid_kill", "session.uuid"); >> >> session.execute("uuid_kill session.uuid"); >> >> but there is no output for the session.execute. >> >> Whether it is possible to execute uuid_kill in javascript >> >> Thanks in advance >> >> >> Warm Regards, >> N.Baskar >> >> * >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/64310a4b/attachment-0002.html From brian at freeswitch.org Wed Jun 17 10:29:56 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 12:29:56 -0500 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> References: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> Message-ID: <9D79D650-31E6-4119-8959-A434CE866C87@freeswitch.org> Thats one way to put it ;) /b On Jun 17, 2009, at 12:26 PM, Diego Viola wrote: > Applications are the ones in mod_dptools and FSAPI are mod_commands > API right? From brian at freeswitch.org Wed Jun 17 11:05:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:05:12 -0500 Subject: [Freeswitch-users] Fwd: [UniMRCP] Open source ASR and TTS plugins References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> Message-ID: <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Guys I'm tossing in $250 dollars of my own money on this ... who is going to pitch in? Arsen and I have been talking about how to accomplish this so we'll have an Open Source Speech Server via MRCP. Who wants to pitch in please paypal brian at freeswitch.org and I can wire it to Arsen. Thanks, Brian Begin forwarded message: > ----- Forwarded Message ---- > From: Arsen Chaloyan > To: UniMRCP > Sent: Wednesday, June 17, 2009 10:57:30 PM > Subject: [UniMRCP] Open source ASR and TTS plugins > > Anybody interested in the development of open source ASR and TTS > plugins for UniMRCP server write me offlist. > > PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ > Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ > > Thanks, > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/62f1371d/attachment-0002.html From bjbrashier at gmail.com Wed Jun 17 11:18:34 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Wed, 17 Jun 2009 11:18:34 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> Message-ID: <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> Well, since what I really need at this time is only about 5 commands of similar complexity to a toggle on something already extant, I've decided to just modify source. I can't imagine that people will be terribly interested in my modifications, but I know I'm interested in being able to stay updated with the current trunk, so I'll have to figure out how to deal with that. I'll let you know if I have trouble. On Tue, Jun 16, 2009 at 5:04 PM, William Suffill wrote: > It depends pretty heavily on what you are trying to add function wise. If > it's more in depth using the event socket would allow it to be used on any > FreeSwitch server assuming it caught the dtmf and acted according without > having to modify the core source code/recompile. It might be a bit more work > at first but could be well worth it depending on your needs. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/42c65616/attachment-0002.html From msc at freeswitch.org Wed Jun 17 11:26:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Jun 2009 13:26:09 -0500 Subject: [Freeswitch-users] javascript session.execute In-Reply-To: <9D79D650-31E6-4119-8959-A434CE866C87@freeswitch.org> References: <191c3a030906170627u3aa818a0oa56b4f621b3ac303@mail.gmail.com> <86a32abc0906171026i528c50b8y8927c73953cb0261@mail.gmail.com> <9D79D650-31E6-4119-8959-A434CE866C87@freeswitch.org> Message-ID: <87f2f3b90906171126l13cabd0atf0aa42a282385d56@mail.gmail.com> Another way to view it: FSAPI or "APIs" are done at the CLI. (API at the CLI) dptools or dial plan applications (apps) are done inside the dialplan So if you have something like "api.execute" then you would use that to execute an FSAPI, just as if you'd typed it at the CLI. On the other hand if you have a session object that's like having a dialplan in your script, so you do dialplan type stuff with a session object. Hope that helps... -MC On Wed, Jun 17, 2009 at 12:29 PM, Brian West wrote: > Thats one way to put it ;) > > /b > > On Jun 17, 2009, at 12:26 PM, Diego Viola wrote: > > > Applications are the ones in mod_dptools and FSAPI are mod_commands > > API right? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/9148f268/attachment-0002.html From shaheryarkh at googlemail.com Wed Jun 17 11:43:36 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 18 Jun 2009 00:43:36 +0600 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia Message-ID: Hi, Is it possible to enable compact SIP headers in mod_sofia configuration? If yes, then how to do so? Kindly give an example. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/6747580f/attachment-0002.html From mitul at enterux.com Wed Jun 17 11:50:52 2009 From: mitul at enterux.com (Mitul Limbani) Date: Thu, 18 Jun 2009 00:20:52 +0530 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: Brain, I can chip in $150 of my own as well, but two things: 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work on it?) 2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 17-Jun-09, at 23:35, Brian West wrote: > Guys I'm tossing in $250 dollars of my own money on this ... who is > going to pitch in? Arsen and I have been talking about how to > accomplish this so we'll have an Open Source Speech Server via MRCP. > > Who wants to pitch in please paypal brian at freeswitch.org and I can > wire it to Arsen. > > Thanks, > Brian > > > Begin forwarded message: > >> ----- Forwarded Message ---- >> From: Arsen Chaloyan >> To: UniMRCP >> Sent: Wednesday, June 17, 2009 10:57:30 PM >> Subject: [UniMRCP] Open source ASR and TTS plugins >> >> Anybody interested in the development of open source ASR and TTS >> plugins for UniMRCP server write me offlist. >> >> PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ >> Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ >> >> Thanks, >> -- >> Arsen Chaloyan >> The author of UniMRCP >> http://www.unimrcp.org > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/8a83c0ef/attachment-0002.html From brian at freeswitch.org Wed Jun 17 11:54:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:54:03 -0500 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia In-Reply-To: References: Message-ID: Its not possible right now but you could if you enable the config option and apply the tag... its something I have thought about adding but wasn't high on my list. NTATAG_SIPFLAGS(MSG_FLG_COMPACT) http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 /b On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: > Hi, > > Is it possible to enable compact SIP headers in mod_sofia > configuration? If yes, then how to do so? Kindly give an example. > > Thank you. From brian at freeswitch.org Wed Jun 17 11:55:19 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:55:19 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: mod_pocketsphinx will be there still as will mod_flite.. this lets you offload ASR and TTS to another server in a standard way. ScribleJ hasn't really helped me with mod_pocketsphinx. The out of grammar segfault is now gone if you update ;) /b On Jun 17, 2009, at 1:50 PM, Mitul Limbani wrote: > Brain, > > I can chip in $150 of my own as well, but two things: > > 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work > on it?) > 2) What are the deliverable we expecting of Arsen's effort? And a > generic ETA of those deliverables? > > Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt Ltd, > The Enterprise Linux Company(r), > http://www.enterux.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/45b407fe/attachment-0002.html From edpimentl at gmail.com Wed Jun 17 11:56:31 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 17 Jun 2009 14:56:31 -0400 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> I will match the 150.00 Best regards, -E CEO and Founder Gpro.ws http://Twitter.com/edpimentl http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://TweetUp.ws (Twitter based MeetUp service) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/f88436a5/attachment-0002.html From brian at freeswitch.org Wed Jun 17 11:59:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 13:59:57 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> Message-ID: <81ACBAA5-DCF6-4475-B225-F12A217D3BA4@freeswitch.org> Thanks for the match. /b On Jun 17, 2009, at 1:56 PM, EdPimentl wrote: > I will match the 150.00 > > Best regards, > -E > CEO and Founder > Gpro.ws > http://Twitter.com/edpimentl > > http://TwebEX.com (Twitter Based Online Web Conference Platform) > http://TwitrShare.com (Send Picture and Message to Tweet Contacts) > http://TweetUp.ws (Twitter based MeetUp service) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/2f4bb10e/attachment-0002.html From kevin at johnnyvoip.com Wed Jun 17 11:57:29 2009 From: kevin at johnnyvoip.com (Kevin Green) Date: Wed, 17 Jun 2009 14:57:29 -0400 Subject: [Freeswitch-users] Solaris 10 - Install Instructions Needed Message-ID: I was wondering if anyone had a set of detailed step-by-step instructions on how to get FS up and running on Solaris 10 from a base Solaris install. I understand that there are a few pieces required to be able to build FS as described on the wiki ( http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris) but it skips over the details of each step. If someone could fill in the details or provide me with some help it would be greatly appreciated. Also, if there are any differences between working on an x86 vs a SPARC platform it would be good to have notes on the different steps that should be taken including any configuration variables that would optomize FS for x86 or SPARC CoolThreads machine. I have attached a whitepaper describing the benefit of UltraSPARC T2 processors running VoIP applications for anyone's interest. This whitepaper is my motivation for wanting to try FS on Solaris. Regards, Kevin Green JohnnyVoIP 350 Legget Drive Kanata, ON, Canada K2K 2W7 Phone: 613 271 5993 Ext 1203 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a6af22ae/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: 02a.SMI.323.SIP_UltraSPARC_WP.pdf Type: application/pdf Size: 145089 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a6af22ae/attachment-0002.pdf From brian at freeswitch.org Wed Jun 17 12:00:10 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 14:00:10 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> Message-ID: <04FB0300-BED1-4B8A-8F95-21D767B3B0C6@freeswitch.org> Btw thanks for the donation! ;) /b On Jun 17, 2009, at 1:50 PM, Mitul Limbani wrote: > Brain, > > I can chip in $150 of my own as well, but two things: > > 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work > on it?) > 2) What are the deliverable we expecting of Arsen's effort? And a > generic ETA of those deliverables? > > Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt Ltd, > The Enterprise Linux Company(r), > http://www.enterux.com/ > > > On 17-Jun-09, at 23:35, Brian West wrote: > >> Guys I'm tossing in $250 dollars of my own money on this ... who is >> going to pitch in? Arsen and I have been talking about how to >> accomplish this so we'll have an Open Source Speech Server via MRCP. >> >> Who wants to pitch in please paypal brian at freeswitch.org and I can >> wire it to Arsen. >> >> Thanks, >> Brian >> >> >> Begin forwarded message: >> >>> ----- Forwarded Message ---- >>> From: Arsen Chaloyan >>> To: UniMRCP >>> Sent: Wednesday, June 17, 2009 10:57:30 PM >>> Subject: [UniMRCP] Open source ASR and TTS plugins >>> >>> Anybody interested in the development of open source ASR and TTS >>> plugins for UniMRCP server write me offlist. >>> >>> PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ >>> Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ >>> >>> Thanks, >>> -- >>> Arsen Chaloyan >>> The author of UniMRCP >>> http://www.unimrcp.org >> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> dev >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/55351f8a/attachment-0002.html From larclap at yahoo.com Wed Jun 17 12:09:26 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 17 Jun 2009 12:09:26 -0700 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? In-Reply-To: <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> References: <012901c9ef6e$01b3ee60$051bcb20$@com> <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> Message-ID: <019901c9ef7f$221498b0$663dca10$@com> Michael, The expression is part of version 13723 distribution in conf/dialplan/default.xml. Shouldn't that be changed? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, June 17, 2009 10:23 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] eavesdrop extension condition in default.xml? On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb wrote: In conf/dialplan/default.xml, the eavesdrop extension's condition is - expression="^88(.*)$|^\*0(.*)$"> Is this intended? I thought it was defined to eavesdrop on internal extensions. Why wouldn't it be something like - expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop extension. Lars, "8885551212" should never match regex expression "^88(\d{3,5})$" . It looks to me like your eavesdrop is matching on "^88(.*)$" which most definitely WILL match 8885551212. Please check your dialplan's eavesdrop extension's regex and make sure it is correct. -MC Log: 1030 Dialplan: sofia/internal/1009 at 192..168.10.29 parsing [default->eavesdrop] continue=false 1031 Dialplan: sofia/internal/1009 at 192..168.10.29 Regex (PASS) [eavesdrop] destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false 1032 Dialplan: sofia/internal/1009 at 192..168.10.29 Action answer() Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/16c1cd04/attachment-0002.html From rupa at rupa.com Wed Jun 17 12:12:19 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 17 Jun 2009 14:12:19 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> Message-ID: I added the ability to call into your dialplan from the caller controls in conferences a while back. Depending on your goal, that might be an easy way to get your problem resolved. You can keep state using the hash api and hash on the conference name or some other useful thingie. On Wed, Jun 17, 2009 at 1:18 PM, Bradley Brashier wrote: > Well, since what I really need at this time is only about 5 commands of > similar complexity to a toggle on something already extant, I've decided to > just modify source. I can't imagine that people will be terribly interested > in my modifications, but I know I'm interested in being able to stay updated > with the current trunk, so I'll have to figure out how to deal with that. > > I'll let you know if I have trouble. > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/83cc02dd/attachment-0002.html From shaheryarkh at googlemail.com Wed Jun 17 12:22:09 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 18 Jun 2009 01:22:09 +0600 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia In-Reply-To: References: Message-ID: Ok, thanks, i will take care of it in my code where necessary. Thank you. On Thu, Jun 18, 2009 at 12:54 AM, Brian West wrote: > Its not possible right now but you could if you enable the config > option and apply the tag... its something I have thought about adding > but wasn't high on my list. > > NTATAG_SIPFLAGS(MSG_FLG_COMPACT) > > > http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 > > /b > > On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: > > > Hi, > > > > Is it possible to enable compact SIP headers in mod_sofia > > configuration? If yes, then how to do so? Kindly give an example. > > > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/9f8f0481/attachment-0002.html From achaloyan at yahoo.com Wed Jun 17 12:56:29 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Wed, 17 Jun 2009 12:56:29 -0700 (PDT) Subject: [Freeswitch-users] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <61DF2924-3589-4DD5-9B2A-6CCF524DB3D2@freeswitch.org> References: <61DF2924-3589-4DD5-9B2A-6CCF524DB3D2@freeswitch.org> Message-ID: <750446.41188.qm@web111302.mail.gq1.yahoo.com> Mitul, >2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Both PocketSphinx and Flite will be integrated into open source UniMRCP server as ASR and TTS plugins respectively and will be publicly available under the Apache 2.0 license as UniMRCP itself. It's matter of 2~3 weeks to have basically working and usable plugins for them. Thank you, Arsen. From: Mitul Limbani Date: June 17, 2009 1:50:52 PM CDT To: "freeswitch-dev at lists.freeswitch.org" Cc: "freeswitch-dev at lists.freeswitch.org" , "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins Reply-To: freeswitch-users at lists.freeswitch.org Brain, I can chip in $150 of my own as well, but two things: 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work on it?) 2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 17-Jun-09, at 23:35, Brian West wrote: Guys I'm tossing in $250 dollars of my own money on this ... who is going to pitch in? Arsen and I have been talking about how to accomplish this so we'll have an Open Source Speech Server via MRCP. Who wants to pitch in please paypal brian at freeswitch.org and I can wire it to Arsen. Thanks, Brian Begin forwarded message: ----- Forwarded Message ---- From: Arsen Chaloyan To: UniMRCP Sent: Wednesday, June 17, 2009 10:57:30 PM Subject: [UniMRCP] Open source ASR and TTS plugins Anybody interested in the development of open source ASR and TTS plugins for UniMRCP server write me offlist. PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ Thanks, -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/154709ba/attachment-0002.html From msc at freeswitch.org Wed Jun 17 13:08:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Jun 2009 15:08:15 -0500 Subject: [Freeswitch-users] eavesdrop extension condition in default.xml? In-Reply-To: <019901c9ef7f$221498b0$663dca10$@com> References: <012901c9ef6e$01b3ee60$051bcb20$@com> <87f2f3b90906171022o215d700ft8ae46345d0145d23@mail.gmail.com> <019901c9ef7f$221498b0$663dca10$@com> Message-ID: <87f2f3b90906171308g73499dccj9520caacdfd73593@mail.gmail.com> On Wed, Jun 17, 2009 at 2:09 PM, Lars Zeb wrote: > Michael, > > > > The expression is part of version 13723 distribution in > conf/dialplan/default.xml. Shouldn?t that be changed? > > > > > > "^88(.*)$|^\*0(.*)$"> > > > > "${hash(select/${domain_name}-spymap/$1)}"/> > > > > > > > Yes, it should be changed. What I'm saying is that the dialplan is matching on the regex expression "^88(.*)$" which will match your toll free numbers. In plain English it means that your dialplan is not correct somewhere. If you've got the new expression of "^88(\d{3,5})$" in your dialplan then that SHOULD work. Go double-check your dialplan and make sure that you don't have the wrong regex in there somewhere. Also, make sure that you reloadxml (or press F6) to make sure that your new changes are all properly loaded. -MC > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, June 17, 2009 10:23 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] eavesdrop extension condition in > default.xml? > > > > > > On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb wrote: > > In conf/dialplan/default.xml, the eavesdrop extension's condition is - > expression="^88(.*)$|^\*0(.*)$"> > > > > Is this intended? I thought it was defined to eavesdrop on internal > extensions. Why wouldn't it be something like - > expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial > 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop > extension. > > > > Lars, > "8885551212" should never match regex expression "^88(\d{3,5})$" . It > looks to me like your eavesdrop is matching on "^88(.*)$" which most > definitely WILL match 8885551212. Please check your dialplan's eavesdrop > extension's regex and make sure it is correct. > > -MC > > Log: > > 1030 Dialplan: sofia/internal/1009 at 192..168.10.29parsing [default->eavesdrop] continue=false > > 1031 Dialplan: sofia/internal/1009 at 192..168.10.29Regex (PASS) [eavesdrop] destination_number(8885819795) =~ > /^88(.*)$|^\*0(.*)$/ break=on-false > > 1032 Dialplan: sofia/internal/1009 at 192..168.10.29Action answer() > > > > Thanks, Lars > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/b1ebcf4d/attachment-0002.html From raul at etellicom.com Wed Jun 17 13:10:32 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 17 Jun 2009 17:10:32 -0300 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090617073606.GA13623@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> Message-ID: <1245269432.16148.22.camel@raul-laptop> I can confirm the same issue, but it happens even with all the IPv6 stuff removed. This is my sofia status: freeswitch at internal> sofia status Name Type Data State ================================================================================================= internal profile sip:mod_sofia at 10.0.13.10:5060 RUNNING (0) imxtony.sytes.net alias internal ALIASED external profile sip:mod_sofia at 10.0.13.10:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG italk2 gateway sip:6499744069 at akl.italk.co.nz REGED italk gateway sip:6499744074 at akl.italk.co.nz REGED ================================================================================================= 2 profiles 1 alias And this is the INVITE for a gateway from an internal endpoint: freeswitch at internal> originate sofia/imxtony.sytes.net/218 &bridge(sofia/gateway/italk/6499744074) ------------------------------------------------------------------------ send 1344 bytes to udp/[203.184.16.2]:5060 at 20:01:50.190193: ------------------------------------------------------------------------ INVITE sip:6499744074 at akl.italk.co.nz SIP/2.0 Via: SIP/2.0/UDP 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np Max-Forwards: 70 From: "FreeSWITCH" ;tag=KSgjD3t33N9yp To: Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 CSeq: 116516120 INVITE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk- at SWITCH_VERSION_REVISION@ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Proxy-Authorization: Digest username="6499744074", realm="italk.co.nz", nonce="75444262", algorithm=MD5, uri="sip:6499744074 at akl.italk.co.nz", response="89bb5673f48252622025641153b882de" Content-Type: application/sdp Content-Disposition: session Content-Length: 315 Remote-Party-ID: "FreeSWITCH" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1245252480 1245252481 IN IP6 stun:stun.freeswitch.org s=FreeSWITCH c=IN IP6 stun:stun.freeswitch.org t=0 0 m=audio 16430 RTP/AVP 0 3 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2009-06-18 08:01:50.191067 [DEBUG] sofia.c:3210 Channel sofia/external/6499744074 entering state [calling][0] recv 469 bytes from udp/[203.184.16.2]:5060 at 20:01:50.252230: ------------------------------------------------------------------------ SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np;received=60.234.179.34 From: "FreeSWITCH" ;tag=KSgjD3t33N9yp To: ;tag=as34133b10 Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 CSeq: 116516120 INVITE User-Agent: italk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ------------------------------------------------------------------------ send 360 bytes to udp/[203.184.16.2]:5060 at 20:01:50.252988: ------------------------------------------------------------------------ ACK sip:6499744074 at akl.italk.co.nz SIP/2.0 Via: SIP/2.0/UDP 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np Max-Forwards: 70 From: "FreeSWITCH" ;tag=KSgjD3t33N9yp To: ;tag=as34133b10 Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 CSeq: 116516120 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-06-18 08:01:50.253252 [DEBUG] sofia.c:3210 Channel sofia/external/6499744074 entering state [terminated][488] That IP6 is clearly messing things up. Regards, Raul On Wed, 2009-06-17 at 17:36 +1000, Jason White wrote: > Jason White wrote: > > > The symptom is the following line in outgoing SIP messages while attempting to > > establish a call to a gateway via the external profile: > > > > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org > > However, if I place an IPv6 call via the internal-ipv6 profile, the o= line > contains the correct IPv6 address - so this is only adversely affecting the > IPv4 external profile, it seems. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/f98483e0/attachment-0002.html From jmesquita at gmail.com Wed Jun 17 13:26:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 17 Jun 2009 17:26:50 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <24067617.post@talk.nabble.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> Message-ID: <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP "tricks" 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: > > Hi, > > look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" > where you can ask for special requirements. > > NOx > > > > Diego Viola wrote: > > > > Hi everyone, > > > > Can you please recommend me some GSM gateway? I'm currently looking > > for a good one to buy... anyone have experience PORTech GSM gateways? > > Are they good? > > > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > > > Thanks, > > > > Diego > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/4a2d570a/attachment-0002.html From apt.get at gmail.com Wed Jun 17 14:15:43 2009 From: apt.get at gmail.com (David Burgess) Date: Wed, 17 Jun 2009 15:15:43 -0600 Subject: [Freeswitch-users] high latency In-Reply-To: References: <87f2f3b90906121533o3c20c10ew503a5919f0c6fa05@mail.gmail.com> <87f2f3b90906121628r75c80bddi2492fcafc5caa3f0@mail.gmail.com> Message-ID: On Fri, Jun 12, 2009 at 5:34 PM, Mathieu Rene wrote: > Try > Math This seems to have greatly reduced but not eliminated late-onset latency :p The latest version of the pfsense-freeswitch package is based on freeswitch 13784 and claims to have eliminated the issue, but I haven't tried it yet. db From jan.kubr at gmail.com Wed Jun 17 14:31:06 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Wed, 17 Jun 2009 23:31:06 +0200 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> Message-ID: <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan 2009/6/17 Jo?o Mesquita > Guys, I was looking at the advantages and disadvantages of having a GSM > gateway vs. a GSM board. > > The conclusions I get are: > > Board pros > > 1. Boards are able to get/send SMS without SIP "tricks" > 2. You don't have to make a SIP call to check if channel is available and > don't rely o SIP messages to get channel status > 3. FS will be able to check for signal level on the board and fire events > on pre-defined thresholds. > > Gateway pros > > 1. I think of is the a GW can be used by more then one server, therefore, > can have failover. > 2. A GW is more scalable > > It would be nice if you, that have already used GSM GWs in production, > could comment on this. > > Thanks, > > jmesquita > > > On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: > >> >> Hi, >> >> look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" >> where you can ask for special requirements. >> >> NOx >> >> >> >> Diego Viola wrote: >> > >> > Hi everyone, >> > >> > Can you please recommend me some GSM gateway? I'm currently looking >> > for a good one to buy... anyone have experience PORTech GSM gateways? >> > Are they good? >> > >> > I also need it to work with FS, I'm also kinda new with VoIP hardware. >> > >> > Thanks, >> > >> > Diego >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/4b731506/attachment-0002.html From jmesquita at gmail.com Wed Jun 17 14:43:46 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 17 Jun 2009 18:43:46 -0300 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> Message-ID: <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> Pricewise, is it worth it? jmesquita On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: > We plan to buy one of these: > > http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html > since you can use SMTP/POP3 to manage SMS. > > Jan > > 2009/6/17 Jo?o Mesquita > > Guys, I was looking at the advantages and disadvantages of having a GSM >> gateway vs. a GSM board. >> >> The conclusions I get are: >> >> Board pros >> >> 1. Boards are able to get/send SMS without SIP "tricks" >> 2. You don't have to make a SIP call to check if channel is available and >> don't rely o SIP messages to get channel status >> 3. FS will be able to check for signal level on the board and fire events >> on pre-defined thresholds. >> >> Gateway pros >> >> 1. I think of is the a GW can be used by more then one server, therefore, >> can have failover. >> 2. A GW is more scalable >> >> It would be nice if you, that have already used GSM GWs in production, >> could comment on this. >> >> Thanks, >> >> jmesquita >> >> >> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >> >>> >>> Hi, >>> >>> look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" >>> where you can ask for special requirements. >>> >>> NOx >>> >>> >>> >>> Diego Viola wrote: >>> > >>> > Hi everyone, >>> > >>> > Can you please recommend me some GSM gateway? I'm currently looking >>> > for a good one to buy... anyone have experience PORTech GSM gateways? >>> > Are they good? >>> > >>> > I also need it to work with FS, I'm also kinda new with VoIP hardware. >>> > >>> > Thanks, >>> > >>> > Diego >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a307bda5/attachment-0002.html From R.Kloosterman at mtel.nl Wed Jun 17 14:44:25 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Wed, 17 Jun 2009 23:44:25 +0200 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com><24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A9AF@srvmtel.office.mtel.nl> I have used board-based voice solutions for ISDN/SS7 over the last 10 years and just moved to gateways and VoIP 2 years ago. I must say I've never been more happy. It scales excellent and is very stable, never had any problem and it's handling about 30.000 calls per day. In the past I've had a fair share of board issues, some crashes, scaling headache. Not totaly bad, but the gateway approach is much nicer. Check out audiocodes, I believe their mediant series supports GSM as well. Remko ________________________________ Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Jo?o Mesquita Verzonden: woensdag 17 juni 2009 22:27 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] Which GSM gateway to buy? Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP "tricks" 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: Hi, look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" where you can ask for special requirements. NOx Diego Viola wrote: > > Hi everyone, > > Can you please recommend me some GSM gateway? I'm currently looking > for a good one to buy... anyone have experience PORTech GSM gateways? > Are they good? > > I also need it to work with FS, I'm also kinda new with VoIP hardware. > > Thanks, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/a612bce2/attachment-0002.html From bjbrashier at gmail.com Wed Jun 17 14:45:52 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Wed, 17 Jun 2009 14:45:52 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> Message-ID: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> So I found one interesting thing so far: the "lock" caller control actually does function as a toggle, and, in fact, "unlock" does not do anything. This goes against wiki docs on mod_conference, but is helpful in this instance. I have a few other commands to work on, still. I found execute-application to be interesting, but since what I next need is a count of all conference participants, and no application already exists for that, I'm still going to have to write something else. BB On Wed, Jun 17, 2009 at 12:12 PM, Rupa Schomaker wrote: > I added the ability to call into your dialplan from the caller controls in > conferences a while back. Depending on your goal, that might be an easy way > to get your problem resolved. You can keep state using the hash api and > hash on the conference name or some other useful thingie. > > On Wed, Jun 17, 2009 at 1:18 PM, Bradley Brashier wrote: > >> Well, since what I really need at this time is only about 5 commands of >> similar complexity to a toggle on something already extant, I've decided to >> just modify source. I can't imagine that people will be terribly interested >> in my modifications, but I know I'm interested in being able to stay updated >> with the current trunk, so I'll have to figure out how to deal with that. >> >> I'll let you know if I have trouble. >> > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/29535a5a/attachment-0002.html From steve at justfone.com Wed Jun 17 14:53:26 2009 From: steve at justfone.com (Steven Brown) Date: Wed, 17 Jun 2009 22:53:26 +0100 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 172 In-Reply-To: References: Message-ID: <3e6d7b0c0906171453y149cbb0n6a4d3522a7495606@mail.gmail.com> You can also send and recieve SMS on the PORTech Gateways via AT commands over a socket connection, I've not checked but would assume all other AT commands will work so you can check signal strength etc this way also. Steve Message: 4 Date: Wed, 17 Jun 2009 23:31:06 +0200 From: Jan Kubr Subject: Re: [Freeswitch-users] Which GSM gateway to buy? To: freeswitch-users at lists.freeswitch.org Message-ID: <698401620906171431t10432015xa78976a401dc5c5b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090617/601ffc4c/attachment-0002.html From jaybinks at gmail.com Wed Jun 17 15:28:29 2009 From: jaybinks at gmail.com (jay binks) Date: Thu, 18 Jun 2009 08:28:29 +1000 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> Message-ID: Ive used these in the past. http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html sound fine, work well... reliable etc etc.. things to watch out for... : * cant send your own caller ID from them ( in my experience its locked to the sim ) * your provider might block the IMEI number of the GSM terminal, if they dont like what your doing. just some stuff to consider. Jay 2009/6/18 Jo?o Mesquita > Pricewise, is it worth it? > > jmesquita > > > On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: > >> We plan to buy one of these: >> >> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >> since you can use SMTP/POP3 to manage SMS. >> >> Jan >> >> 2009/6/17 Jo?o Mesquita >> >> Guys, I was looking at the advantages and disadvantages of having a GSM >>> gateway vs. a GSM board. >>> >>> The conclusions I get are: >>> >>> Board pros >>> >>> 1. Boards are able to get/send SMS without SIP "tricks" >>> 2. You don't have to make a SIP call to check if channel is available and >>> don't rely o SIP messages to get channel status >>> 3. FS will be able to check for signal level on the board and fire events >>> on pre-defined thresholds. >>> >>> Gateway pros >>> >>> 1. I think of is the a GW can be used by more then one server, therefore, >>> can have failover. >>> 2. A GW is more scalable >>> >>> It would be nice if you, that have already used GSM GWs in production, >>> could comment on this. >>> >>> Thanks, >>> >>> jmesquita >>> >>> >>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >>> >>>> >>>> Hi, >>>> >>>> look at www.kuhnt.com. It?s a german page. There you can find "Kontakt" >>>> where you can ask for special requirements. >>>> >>>> NOx >>>> >>>> >>>> >>>> Diego Viola wrote: >>>> > >>>> > Hi everyone, >>>> > >>>> > Can you please recommend me some GSM gateway? I'm currently looking >>>> > for a good one to buy... anyone have experience PORTech GSM gateways? >>>> > Are they good? >>>> > >>>> > I also need it to work with FS, I'm also kinda new with VoIP hardware. >>>> > >>>> > Thanks, >>>> > >>>> > Diego >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/b5eec936/attachment-0002.html From jason at jasonjgw.net Wed Jun 17 18:01:30 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 18 Jun 2009 11:01:30 +1000 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <1245269432.16148.22.camel@raul-laptop> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> Message-ID: <20090618010130.GA14634@jdc.jasonjgw.net> Raul Fragoso wrote: > I can confirm the same issue, but it happens even with all the IPv6 > stuff removed. Thank you for the corroboration. It only happens to me if I have the following in my external.xml profile: Note that I need the above line for nat traversal; if I leave it out, FreeSWITCH uses a private IPv4 address in outgoing invites, and can't establish a call. From jingwei.yang at gmail.com Wed Jun 17 19:50:35 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 18 Jun 2009 10:50:35 +0800 Subject: [Freeswitch-users] Segmentation fault with record_session In-Reply-To: <191c3a030906170618v214fb0e9y4d5c64b724a97200@mail.gmail.com> References: <13529f9d0906150240q6783bee1yc4ebd4980916a896@mail.gmail.com> <7b197bef0906150516p68b1ac0eg6ca25377d028d90e@mail.gmail.com> <13529f9d0906160115s3b04a4b7i2d4f8e5cefa871a3@mail.gmail.com> <7b197bef0906160142n640c3d2cuf1df994abc62aad4@mail.gmail.com> <13529f9d0906160232k2ad22be5s3615f7e7d3715837@mail.gmail.com> <13529f9d0906170239o39bd2c52pa6800d7d88878333@mail.gmail.com> <191c3a030906170618v214fb0e9y4d5c64b724a97200@mail.gmail.com> Message-ID: <13529f9d0906171950j26df5790r19e0ccd8328dcd4d@mail.gmail.com> Hi Anthony, I've filed the report earlier on. This is the url of the bug: http://jira.freeswitch.org/browse/MODSKYPIAX-35. Hi Giovanni, Since the work is still under the development phase, I only managed to get 8 concurrent calls. Here's the cpu and memory consumption data: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 3822 root 15 0 123m 39m 10m S 8.6 1.0 0:37.60 skype 4801 root 18 0 48304 20m 5352 S 4.0 0.5 0:12.24 freeswitch 3862 root 15 0 122m 37m 10m S 3.3 0.9 0:18.73 skype 3842 root 15 0 120m 36m 10m S 2.3 0.9 0:26.36 skype 3882 root 15 0 121m 37m 10m S 2.3 0.9 0:20.63 skype 3922 root 15 0 98.0m 33m 9604 S 2.3 0.8 0:12.77 skype 3902 root 15 0 120m 36m 10m S 2.0 0.9 0:15.89 skype 3942 root 15 0 99.9m 34m 9628 S 1.3 0.9 0:12.05 skype 4065 root 15 0 71272 33m 9180 S 0.7 0.8 0:06.50 skype 4100 root 15 0 14060 8708 1804 S 0.7 0.2 0:04.55 Xvfb Regards, -Jingwei On Wed, Jun 17, 2009 at 9:18 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You should not report bugs on the mailing list. > please report your bug to jira http://jira.freeswitch.org > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Make sure you attach a backtrace of your issue and file it under skypeiax > so giovanni can track it. > > > On Wed, Jun 17, 2009 at 4:39 AM, Jingwei Yang wrote: > >> Hi Giovanni, >> >> Sorry, pretty busy and fully occupied by other stuff today. Have to delay >> the testing and give you the result tomorrow. >> >> Regards, >> -Jingwei >> >> >> On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang wrote: >> >>> Sure, I'll append to you the result tomorrow. >>> >>> Regards, >>> -Jingwei >>> >>> >>> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli < >>> gmaruzz at celliax.org> wrote: >>> >>>> Hi Jingwei, >>>> >>>> Thanks a lot! I'll take care of as soon as possible. >>>> >>>> Btw, before I read the Jira, are you testing in linux? >>>> >>>> If you are testing on linux, would you please report how it is >>>> performing under load? I mean, what is the load average with, let say, >>>> 10 or 20 or more concurrent Skype call? >>>> >>>> This has nothing to do with your bug, but will help me in getting >>>> better performances. >>>> >>>> Ciao for now, and thanks again for reporting! >>>> >>>> -giovanni >>>> >>>> >>>> >>>> >>>> On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yang >>>> wrote: >>>> > Hi Giovanni, >>>> > >>>> > I've reported it in Jira. Here's the bug url: >>>> > >>>> > http://jira.freeswitch.org/browse/MODSKYPIAX-35 >>>> > >>>> > Thanks, >>>> > -Jingwei >>>> > >>>> > On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli < >>>> gmaruzz at celliax.org> >>>> > wrote: >>>> >> >>>> >> Hi Jingwel, >>>> >> thanks for reporting. >>>> >> >>>> >> Could you please add a Jira issue with as much details as possible? >>>> >> >>>> >> general guide for reporting bugs: >>>> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >> >>>> >> what to add for skypiax: >>>> >> >>>> >> >>>> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests >>>> >> >>>> >> mod_skypiax Jira: >>>> >> http://jira.freeswitch.org/browse/MODSKYPIAX >>>> >> >>>> >> >>>> >> Sincerely, >>>> >> >>>> >> Giovanni Maruzzelli >>>> >> ========================================= >>>> >> www.celliax.org >>>> >> via Pierlombardo 9, 20135 Milano >>>> >> Italy >>>> >> gmaruzz at celliax dot org >>>> >> Cell : +39-347-2665618 >>>> >> Fax : +39-02-87390039 >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yang< >>>> jingwei.yang at gmail.com> >>>> >> wrote: >>>> >> > Hi Team, >>>> >> > >>>> >> > I've been using the record_session feature to record call sessions. >>>> >> > Here's >>>> >> > how I prepared the dialplan: >>>> >> > >>>> >> > >>>> >> > >>> expression="^2909/(.*)$"> >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > And here's how I trigger it: >>>> >> > >>>> >> > freeswitch at localhost.localdomain>originate >>>> skypiax/skypiax2/userAAA >>>> >> > 2909/userBBB >>>> >> > >>>> >> > The call can be established and the data.wav file was generated >>>> without >>>> >> > any >>>> >> > problem. However, once userAAA hung up, a segmentation fault >>>> occurred >>>> >> > and >>>> >> > freeswitch was automatically shut down. Here are what I got from >>>> the >>>> >> > console: >>>> >> > >>>> >> > freeswitch at localhost.localdomain> originate >>>> skypiax/skypiax2/userAAA >>>> >> > 2909/userBBB >>>> >> > 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 >>>> >> > switch_channel_set_name() >>>> >> > New Channel skypiax/skypiax2/userAAA >>>> >> > [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] >>>> >> > 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 >>>> >> > remote_party_is_ringing() >>>> >> > Ring-Ready skypiax/skypiax2/userAAA >>>> >> > 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 >>>> >> > outbound_channel_answered() >>>> >> > Channel [skypiax/skypiax2/userAAA] has been answered >>>> >> > 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 >>>> >> > switch_ivr_session_transfer() >>>> >> > Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB at default] >>>> >> > API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: >>>> >> > +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b >>>> >> > >>>> >> > freeswitch at localhost.localdomain> 2009-06-15 17:25:10 [INFO] >>>> >> > mod_dialplan_xml.c:252 dialplan_hunt() Processing >>>> >> > FreeSWITCH->2909/userBBB >>>> >> > in context default >>>> >> > 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 >>>> >> > switch_channel_set_name() >>>> >> > New Channel skypiax/ANY/userBBB >>>> [4a8b36a4-85d6-4735-98df-dde1a32ac66a] >>>> >> > 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 >>>> >> > remote_party_is_ringing() >>>> >> > Ring-Ready skypiax/ANY/userBBB! >>>> >> > 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 >>>> >> > outbound_channel_answered() >>>> >> > Channel [skypiax/ANY/userBBB] has been answered >>>> >> > 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 >>>> >> > skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA >>>> >> > [CS_EXECUTE] >>>> >> > [NORMAL_CLEARING] >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 >>>> >> > audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB >>>> >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>>> >> > switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) >>>> Ended >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>>> >> > switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA >>>> >> > [CS_DESTROY] >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 >>>> >> > switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended >>>> >> > 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 >>>> >> > switch_core_session_thread() Close Channel skypiax/ANY/userBBB >>>> >> > [CS_DESTROY] >>>> >> > Segmentation fault (core dumped) >>>> >> > >>>> >> > Please kindly let me know whether there's anything wrong with the >>>> >> > dialplan >>>> >> > or the way how I originated the call. >>>> >> > >>>> >> > Thanks! >>>> >> > -Jingwei >>>> >> > >>>> >> > >>>> >> > _______________________________________________ >>>> >> > Freeswitch-users mailing list >>>> >> > Freeswitch-users at lists.freeswitch.org >>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> > http://www.freeswitch.org >>>> >> > >>>> >> > >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/1a435bd5/attachment-0002.html From j3flight at gmail.com Wed Jun 17 17:13:56 2009 From: j3flight at gmail.com (j3flight) Date: Wed, 17 Jun 2009 17:13:56 -0700 (PDT) Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> Message-ID: <24084409.post@talk.nabble.com> I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... Here's a wiki page I created after building a JavaScript IVR for a conference server... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I believe you could stick those in a script by themselves and call them from execute_application. Somehow, you would have to identify what user is calling the script and what conference they're in. (You could possibly set a session variable upon entering the conference, or parse all the conferences until you find that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! -- View this message in context: http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24084409.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From j3flight at gmail.com Wed Jun 17 17:51:13 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Wed, 17 Jun 2009 19:51:13 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> Message-ID: <4A398F81.1020303@gmail.com> I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... Here's a wiki page I created after building a JavaScript IVR for a conference server... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I believe you could stick those in a script by themselves and call them from execute_application. Somehow, you would have to identify what user is calling the script and what conference they're in. (Once you're in the javascript, you could check a "conference number" variable that you set when the person entered the conference. Or, you could parse the output of "conference list" until you found that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! From jcromes at gmail.com Wed Jun 17 20:10:41 2009 From: jcromes at gmail.com (Jason Cromes) Date: Wed, 17 Jun 2009 22:10:41 -0500 Subject: [Freeswitch-users] Controlling Conference Controls Message-ID: I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... This Wiki page has some good JavaScript examples... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I *believe* you could stick those in a script by themselves and call them using execute_application out of the caller controls... Somehow, you would have to identify what user is calling the script and what conference they're in.? (Once you're in the javascript, you could check a "conference number" variable that you set when the person entered the conference.? Or, you could parse the output of "conference list" until you found that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested! From j3flight at gmail.com Wed Jun 17 21:07:28 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Wed, 17 Jun 2009 23:07:28 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> Message-ID: <4A39BD80.8020706@gmail.com> FYI: I fixed the Wiki documentation for the lock/unlock feature. Bradley Brashier wrote: > So I found one interesting thing so far: the "lock" caller control > actually does function as a toggle, and, in fact, "unlock" does not do > anything. This goes against wiki docs on mod_conference, but is > helpful in this instance. From brian at freeswitch.org Wed Jun 17 21:57:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 17 Jun 2009 23:57:03 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <20090618010130.GA14634@jdc.jasonjgw.net> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> <20090618010130.GA14634@jdc.jasonjgw.net> Message-ID: <8B621D34-D0D9-4AAE-B010-36B56DFBBE4C@freeswitch.org> I need in one of your boxs... there is no way this is doing this unless you are putting stun:stun.freeswitch.org into the ext-rtp-ip or sip ip... which could make it trigger the ipv6 check since its just looking for : in the ip address. And stun: has that.. so you're triggering it... tar up your entire conf folder and mail it to me ASAP. /b On Jun 17, 2009, at 8:01 PM, Jason White wrote: > Raul Fragoso wrote: >> I can confirm the same issue, but it happens even with all the IPv6 >> stuff removed. > > Thank you for the corroboration. > > It only happens to me if I have the following in my external.xml > profile: > > > Note that I need the above line for nat traversal; if I leave it out, > FreeSWITCH uses a private IPv4 address in outgoing invites, and can't > establish a call. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Jun 17 22:04:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 00:04:16 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <1245269432.16148.22.camel@raul-laptop> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> Message-ID: <8D2072FB-72BA-4FF2-88E1-C80E200DAD63@freeswitch.org> Ok you both didn't notice you CAN NOT put stun:stun.freeswitch.org in rtp-ip, thats the problem. It clearly says IP ADDRESSES ONLY in the comments. DO not use $${external_rtp_ip} for rtp-ip either :P /b On Jun 17, 2009, at 3:10 PM, Raul Fragoso wrote: > I can confirm the same issue, but it happens even with all the IPv6 > stuff removed. > This is my sofia status: > > freeswitch at internal> sofia status > > Name Type Data > State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > internal profile sip:mod_sofia at 10.0.13.10:5060 > RUNNING (0) > imxtony.sytes.net alias internal > ALIASED > external profile sip:mod_sofia at 10.0.13.10:5080 > RUNNING (0) > example.com gateway sip:joeuser at example.com > NOREG > italk2 gateway sip:6499744069 at akl.italk.co.nz > REGED > italk gateway sip:6499744074 at akl.italk.co.nz > REGED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 2 profiles 1 alias > > And this is the INVITE for a gateway from an internal endpoint: > > freeswitch at internal> originate sofia/imxtony.sytes.net/218 > &bridge(sofia/gateway/italk/6499744074) > > ------------------------------------------------------------------------ > send 1344 bytes to udp/[203.184.16.2]:5060 at 20:01:50.190193: > > ------------------------------------------------------------------------ > INVITE sip:6499744074 at akl.italk.co.nz SIP/2.0 > Via: SIP/2.0/UDP > 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np > Max-Forwards: 70 > From: "FreeSWITCH" 6499744074 at akl.italk.co.nz;transport=udp>;tag=KSgjD3t33N9yp > To: > Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 > CSeq: 116516120 INVITE > Contact: > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk- > @SWITCH_VERSION_REVISION@ > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Proxy-Authorization: Digest username="6499744074", > realm="italk.co.nz", nonce="75444262", algorithm=MD5, uri="sip:6499744074 at akl.italk.co.nz > ", response="89bb5673f48252622025641153b882de" > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 315 > Remote-Party-ID: "FreeSWITCH" 0000000000 at akl.italk.co.nz>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1245252480 1245252481 IN IP6 stun:stun.freeswitch.org > s=FreeSWITCH > c=IN IP6 stun:stun.freeswitch.org > t=0 0 > m=audio 16430 RTP/AVP 0 3 8 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > 2009-06-18 08:01:50.191067 [DEBUG] sofia.c:3210 Channel sofia/ > external/6499744074 entering state [calling][0] > recv 469 bytes from udp/[203.184.16.2]:5060 at 20:01:50.252230: > > ------------------------------------------------------------------------ > SIP/2.0 488 Not acceptable here > Via: SIP/2.0/UDP > 60.234.179.34 > :5080;rport;branch=z9hG4bKgypXrUy0889Np;received=60.234.179.34 > From: "FreeSWITCH" 6499744074 at akl.italk.co.nz;transport=udp>;tag=KSgjD3t33N9yp > To: ;tag=as34133b10 > Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 > CSeq: 116516120 INVITE > User-Agent: italk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > ------------------------------------------------------------------------ > send 360 bytes to udp/[203.184.16.2]:5060 at 20:01:50.252988: > > ------------------------------------------------------------------------ > ACK sip:6499744074 at akl.italk.co.nz SIP/2.0 > Via: SIP/2.0/UDP > 60.234.179.34:5080;rport;branch=z9hG4bKgypXrUy0889Np > Max-Forwards: 70 > From: "FreeSWITCH" 6499744074 at akl.italk.co.nz;transport=udp>;tag=KSgjD3t33N9yp > To: ;tag=as34133b10 > Call-ID: 893ed581-d61c-122c-d885-00237dc8dc02 > CSeq: 116516120 ACK > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2009-06-18 08:01:50.253252 [DEBUG] sofia.c:3210 Channel sofia/ > external/6499744074 entering state [terminated][488] > > > That IP6 is clearly messing things up. > > Regards, > > Raul > > On Wed, 2009-06-17 at 17:36 +1000, Jason White wrote: >> >> Jason White wrote: >> >> > The symptom is the following line in outgoing SIP messages while >> attempting to >> > establish a call to a gateway via the external profile: >> > >> > o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org >> >> However, if I place an IPv6 call via the internal-ipv6 profile, the >> o= line >> contains the correct IPv6 address - so this is only adversely >> affecting the >> IPv4 external profile, it seems. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5b552778/attachment-0002.html From brian at freeswitch.org Wed Jun 17 22:09:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 00:09:44 -0500 Subject: [Freeswitch-users] SIP and nat traversal IPv4/IPv6 issue In-Reply-To: <1245269432.16148.22.camel@raul-laptop> References: <20090617070019.GA5042@jdc.jasonjgw.net> <20090617073606.GA13623@jdc.jasonjgw.net> <1245269432.16148.22.camel@raul-laptop> Message-ID: <93A6FD81-394C-492A-894D-49C8D4CCB1AB@freeswitch.org> I no longer need your configs... I didn't try to put stun:stun.freeswitch.org in sip-ip or rtp-ip because I know you shouldn't. We clearly can not try to do a stun request in either of these fields because you can't bind to IP's that aren't directly on the machine... so do as per the config and put ONLY ip's that are local on the machine... if you need to set the external rtp or sip ip please set them with the ext-rtp-ip and ext-sip-ip params on the profile. You CAN set the ext-rtp-ip and ext-sip-ip to stun:stun.freeswitch.org and those will resolve. /b On Jun 17, 2009, at 3:10 PM, Raul Fragoso wrote: > I can confirm the same issue, but it happens even with all the IPv6 > stuff removed. > This is my sofia status: From darklion11 at yahoo.com Wed Jun 17 22:10:43 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 17 Jun 2009 22:10:43 -0700 (PDT) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> Message-ID: <24086477.post@talk.nabble.com> Not working... CALL Rejected dujinfang wrote: > > comment lines in the user directory do the trick: > > > > > > > > On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote: > >> >> If FS A has an account 8011105 does FS B also nid to register >> 8011105? Yes it >> working on a gateway but the username of the gateway was shown on my >> softphone and also it nids a password for the gateway... is there an >> option >> to view the caller name and number of the FS A gateway to FS B? >> >> >> >> >> Brian West-3 wrote: >>> >>> COPY paste fail :) >>> >>> >>> >>> something like that as per the example. >>> >>> /b >>> >>> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote: >>> >>>> >>>> How can sofia profile can call ACL? >>>> Can you give me an example? >>>> Like this? >>>> >>>> I put this on external profile >>>> >>>> "/> >>>> "/> >>>> >>>> >>>> Brian West-3 wrote: >>>>> >>>>> Now you have to tell the sofia profile to use that ACL >>>>> >>>>> /b >>>>> >>>>> On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote: >>>>> >>>>>> How can i turn off authentication? This is my acl.conf.xml on >>>>>> 192.168.0.105 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 192.168.0.4 >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066647.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24066825.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24086477.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Jun 17 22:14:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 00:14:12 -0500 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24086477.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24086477.post@talk.nabble.com> Message-ID: You're trying way too hard. CALL Rejected gives us exactly ZERO to go on... We are all trying really hard to help you but at some point we just can't help anymore. Please make sure you post debug logs to pastebin and join IRC. This email back and forth over something like this just takes way too long and frustrates everyone... frustration leads to trying to hard... which leads to failure. /b On Jun 18, 2009, at 12:10 AM, Edmar Cruz wrote: > > Not working... CALL Rejected From christian.bourke1 at gmail.com Wed Jun 17 23:03:21 2009 From: christian.bourke1 at gmail.com (Bilbo) Date: Wed, 17 Jun 2009 23:03:21 -0700 (PDT) Subject: [Freeswitch-users] Use Freeswitch to provide a SIP Video/Voice email service Message-ID: <24086865.post@talk.nabble.com> Hi, I would like to use Freeswitch to provide a Video/Voice mail service that is integrated with an email service. I would like to have the ability to email the Video/Voice messages as well as the SIP users being able to collect their Video messages using their video soft-phones. Has anyone done this before or know if Freeswitch is capable? Thanks -- View this message in context: http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-service-tp24086865p24086865.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Wed Jun 17 23:54:29 2009 From: dujinfang at gmail.com (seven) Date: Thu, 18 Jun 2009 14:54:29 +0800 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <24086477.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <24065535.post@talk.nabble.com> <929908.28142.qm@web57310.mail.re1.yahoo.com> <24066647.post@talk.nabble.com> <93456DA1-47C0-4524-903B-0FDE310EE93D@freeswitch.org> <24066825.post@talk.nabble.com> <24086477.post@talk.nabble.com> Message-ID: Note I was saying your caller id problem, how did you see the undesired caller id when you got CALL Rejected? On Jun 18, 2009, at 1:10 PM, Edmar Cruz wrote: > > Not working... CALL Rejected > > dujinfang wrote: >> >> comment lines in the user directory do the trick: >> >> >> >> >> >> >> >> On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote: >> >>> >>> If FS A has an account 8011105 does FS B also nid to register >>> 8011105? Yes it >>> working on a gateway but the username of the gateway was shown on my >>> softphone and also it nids a password for the gateway... is there an >>> option >>> to view the caller name and number of the FS A gateway to FS B? >>> >>> >>> From peter.olsson at visionutveckling.se Thu Jun 18 00:07:22 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 18 Jun 2009 09:07:22 +0200 Subject: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a "api uuid_record start ", and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/d07afbd7/attachment-0002.html From rdenert at tng.de Thu Jun 18 00:17:29 2009 From: rdenert at tng.de (Rudolf Denert) Date: Thu, 18 Jun 2009 09:17:29 +0200 (CEST) Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <18011901.31411245309252614.JavaMail.root@zimbra.tng.de> Message-ID: <2557047.31431245309449504.JavaMail.root@zimbra.tng.de> Yes, I removed the tags but with no effect. I think the problem is that the webserver doesn't look in the directory where featuers.xml is deposited on the freeswitchserver (/opt/freeswitch/conf/dialplan/). The issue is that FS finds the context when dialplan is the directory /opt/freeswitch/conf/dialplan/public/ . But when it is on the webserver (it is on another server with a different IP-address) I get the error that I told you. What should i verify in the default config. Greetz ----- Urspr?ngliche Mail ----- Von: "Brian West" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Freeswitch / Webserver Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > Context features not found _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert at tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. From dujinfang at gmail.com Thu Jun 18 00:20:22 2009 From: dujinfang at gmail.com (seven) Date: Thu, 18 Jun 2009 15:20:22 +0800 Subject: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> Message-ID: <7E13F947-F283-44AD-B718-429632E8319D@gmail.com> can you try uuid_record stop before playback? On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote: > I?m not quite sure if this is the expected behaviour, I just wanted > to make sure. > > I?ve developed a simple IVR application using event socket. I dial > in to the dialplan and park the call, and then I let the IVR > application do whatever it?s supposed to. I basically listen for > DTMF events and play and record files. > > Today I just noticed that if I issue a ?api uuid_record start > ?, and then do a file playback (using SendMsg, with call- > command execute and execute-app-name playback), the playback is sent > both to the caller, and to the recorded file. Is this the way it?s > supposed to work, or should I playback files in another way? > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/88bc875b/attachment-0002.html From d at d-man.org Thu Jun 18 00:24:20 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 18 Jun 2009 00:24:20 -0700 Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: <24065638.post@talk.nabble.com> References: <24065638.post@talk.nabble.com> Message-ID: Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn of "tcapi" still in the config file. If your test was: # isql zenoss edmar edmar Then zenoss should be your db_dsn: Not You should be seeing something about the ODBC connection failing at FreeSWITCH startup if you look at the log closely (search for mod_nibblebill) that indicates this, too. - Darren -----Original Message----- From: Edmar Cruz [mailto:darklion11 at yahoo.com] Sent: Tuesday, June 16, 2009 6:44 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC my nibble.conf.xml param name="db_username" value="edmar"/> param name="db_password" value="edmar"/> param name="db_dsn" value="tcapi"/> param name="db_column_cash" value="cash"/> param name="db_column_account" value="id"/> param name="global_heartbeat" value="1"/> !-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. --> param name="lowbal_amt" value="5"/> param name="lowbal_action" value="play ding"/> param name="nobal_amt" value="0"/> param name="nobal_action" value="hangup"/> param name="percall_max_amt" value="100"/> param name="percall_action" value="hangup"/> Account 1001.xml param name="password" value="1234"/> param name="vm-password" value="1001"/> param name="vm-mailto" value=""/> param name="vm-email-all-messages" value="false"/> param name="vm-delete-file" value="false"/> param name="vm-attach-file" value="false"/> I check unixodbc has been installed. # isql zenoss edmar edmar [SQL]> Connected successfully but on freeswitch error Cannot connect to user ODBC [root] Darren Schreiber wrote: > > What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the > real logs from FS's logs? The info below is not nearly detailed enough. > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Monday, June 15, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC > > > Hi > > I experiencing an error on mod_nibblebill. I already load it from > autoload_configs, especially mod_spidermonkey. Uncomment > mod_spidermonkey_odbc. I also download unixodbc and created the files > /etc/odbcinst.ini and /etc/odbc.ini with the correct format > > [zenoss] > DATABASE = tcapi > USER = root > PASS = password > ..... > > I type also on the console isql zenoss root password. Also working... > > But an error occur on freeswitch Cannot connect to user [root] ... > > What do you thinks is the problem? > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24045890.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 890p24065638.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From peter.olsson at visionutveckling.se Thu Jun 18 00:39:35 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 18 Jun 2009 09:39:35 +0200 Subject: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? In-Reply-To: <7E13F947-F283-44AD-B718-429632E8319D@gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB945D@cooper> <7E13F947-F283-44AD-B718-429632E8319D@gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEDB947B@cooper> Yes I guess this would probably solve the issue :) But since I stumbled across this weird behaviour I just wanted to make sure if this was expected or not, or if it might be a bug... I thought playback was just sending the audio to the caller, but in this case it seems that playback sends it to both "parties". /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r seven Skickat: den 18 juni 2009 09:20 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Weird (or expected) behaviour on record/playback using ESL? can you try uuid_record stop before playback? On Jun 18, 2009, at 3:07 PM, Peter Olsson wrote: I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a "api uuid_record start ", and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a39ebaa32936831919445! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/e9b8f6a3/attachment-0002.html From darklion11 at yahoo.com Thu Jun 18 00:58:36 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 18 Jun 2009 00:58:36 -0700 (PDT) Subject: [Freeswitch-users] Is freeswitch can call mobile phones? Message-ID: <24088222.post@talk.nabble.com> Hi is there any possible free sites ip that i can connect so I can could to any mobiles phones? I know some several ip sites has the capability to call for free Ip to Voip... I know freeswitch can do this Can you give me an example site? -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jan.kubr at gmail.com Thu Jun 18 01:26:26 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Thu, 18 Jun 2009 10:26:26 +0200 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> Message-ID: <698401620906180126n454589a7y3bc2ff4ae7004741@mail.gmail.com> There are gateways that allow you to set your own caller ID? I thought it'd always use the number of the SIM. Jan On Thu, Jun 18, 2009 at 12:28 AM, jay binks wrote: > Ive used these in the past. > > http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html > > sound fine, work well... > reliable etc etc.. > > things to watch out for... : > > * cant send your own caller ID from them ( in my experience its locked to > the sim ) > * your provider might block the IMEI number of the GSM terminal, if they > dont like what your doing. > > just some stuff to consider. > > > Jay > > > > > 2009/6/18 Jo?o Mesquita > > Pricewise, is it worth it? >> >> jmesquita >> >> >> On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: >> >>> We plan to buy one of these: >>> >>> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >>> since you can use SMTP/POP3 to manage SMS. >>> >>> Jan >>> >>> 2009/6/17 Jo?o Mesquita >>> >>> Guys, I was looking at the advantages and disadvantages of having a GSM >>>> gateway vs. a GSM board. >>>> >>>> The conclusions I get are: >>>> >>>> Board pros >>>> >>>> 1. Boards are able to get/send SMS without SIP "tricks" >>>> 2. You don't have to make a SIP call to check if channel is available >>>> and don't rely o SIP messages to get channel status >>>> 3. FS will be able to check for signal level on the board and fire >>>> events on pre-defined thresholds. >>>> >>>> Gateway pros >>>> >>>> 1. I think of is the a GW can be used by more then one server, >>>> therefore, can have failover. >>>> 2. A GW is more scalable >>>> >>>> It would be nice if you, that have already used GSM GWs in production, >>>> could comment on this. >>>> >>>> Thanks, >>>> >>>> jmesquita >>>> >>>> >>>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >>>> >>>>> >>>>> Hi, >>>>> >>>>> look at www.kuhnt.com. It?s a german page. There you can find >>>>> "Kontakt" >>>>> where you can ask for special requirements. >>>>> >>>>> NOx >>>>> >>>>> >>>>> >>>>> Diego Viola wrote: >>>>> > >>>>> > Hi everyone, >>>>> > >>>>> > Can you please recommend me some GSM gateway? I'm currently looking >>>>> > for a good one to buy... anyone have experience PORTech GSM gateways? >>>>> > Are they good? >>>>> > >>>>> > I also need it to work with FS, I'm also kinda new with VoIP >>>>> hardware. >>>>> > >>>>> > Thanks, >>>>> > >>>>> > Diego >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/f0d9a6aa/attachment-0002.html From darklion11 at yahoo.com Thu Jun 18 01:31:22 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 18 Jun 2009 01:31:22 -0700 (PDT) Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to ODBC In-Reply-To: References: <24045890.post@talk.nabble.com> <24065638.post@talk.nabble.com> Message-ID: <24088636.post@talk.nabble.com> Ok thanks a lot for that. Sorry my mistake.. Darren Schreiber wrote: > > Call me crazy but your DB DSN appears to be zenoss, but you have a db dsn > of > "tcapi" still in the config file. > > If your test was: > # isql zenoss edmar edmar > > > Then zenoss should be your db_dsn: > > > Not > > > > You should be seeing something about the ODBC connection failing at > FreeSWITCH startup if you look at the log closely (search for > mod_nibblebill) that indicates this, too. > > - Darren > > > -----Original Message----- > From: Edmar Cruz [mailto:darklion11 at yahoo.com] > Sent: Tuesday, June 16, 2009 6:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error on mod_nibblebill cannot connect to > ODBC > > > my nibble.conf.xml > > > > > > > param name="db_username" value="edmar"/> > param name="db_password" value="edmar"/> > param name="db_dsn" value="tcapi"/> > > > > > > param name="db_column_cash" value="cash"/> > > > param name="db_column_account" value="id"/> > > > > param name="global_heartbeat" value="1"/> > > !-- By default, warn a caller when their balance is at $5.00. You can > set this to a negative number. --> > param name="lowbal_amt" value="5"/> > param name="lowbal_action" value="play ding"/> > > > param name="nobal_amt" value="0"/> > param name="nobal_action" value="hangup"/> > > > param name="percall_max_amt" value="100"/> > param name="percall_action" value="hangup"/> > > > > > Account 1001.xml > > > > > param name="password" value="1234"/> > param name="vm-password" value="1001"/> > param name="vm-mailto" value=""/> > param name="vm-email-all-messages" value="false"/> > param name="vm-delete-file" value="false"/> > param name="vm-attach-file" value="false"/> > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > I check unixodbc has been installed. > > # isql zenoss edmar edmar > [SQL]> > > Connected successfully but on freeswitch error Cannot connect to user ODBC > [root] > > > Darren Schreiber wrote: >> >> What is in your ODBC settings in nibblebill.conf.xml ? Can you paste the >> real logs from FS's logs? The info below is not nearly detailed enough. >> >> -----Original Message----- >> From: Edmar Cruz [mailto:darklion11 at yahoo.com] >> Sent: Monday, June 15, 2009 6:44 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Error on mod_nibblebill cannot connect to >> ODBC >> >> >> Hi >> >> I experiencing an error on mod_nibblebill. I already load it from >> autoload_configs, especially mod_spidermonkey. Uncomment >> mod_spidermonkey_odbc. I also download unixodbc and created the files >> /etc/odbcinst.ini and /etc/odbc.ini with the correct format >> >> [zenoss] >> DATABASE = tcapi >> USER = root >> PASS = password >> ..... >> >> I type also on the console isql zenoss root password. Also working... >> >> But an error occur on freeswitch Cannot connect to user [root] ... >> >> What do you thinks is the problem? >> -- >> View this message in context: >> > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 >> 890p24045890.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045 > 890p24065638.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp24045890p24088636.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From panselva at gmail.com Thu Jun 18 03:06:44 2009 From: panselva at gmail.com (selva kumar) Date: Thu, 18 Jun 2009 15:36:44 +0530 Subject: [Freeswitch-users] Automatic call distribution Message-ID: <45f609f90906180306n2eec6ed3lb8cf84177faf2791@mail.gmail.com> Hi, I have setup FS for both inbound and outbound.It is working fine. Now I would like to configure Automatic Call Distribution(ACD).How to configure it in Freeswitch? Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/2c2964e7/attachment-0002.html From steve at justfone.com Thu Jun 18 06:00:58 2009 From: steve at justfone.com (Steven Brown) Date: Thu, 18 Jun 2009 14:00:58 +0100 Subject: [Freeswitch-users] VAD, TALK and NOTALK events Message-ID: <3e6d7b0c0906180600q1b5f5b3dp4e28350de7aede98@mail.gmail.com> Hi, I have been trying to pick up TALK and NOTALK events but with no success, I have enabled VAD for "both" in my config and the rtp is stopping and starting as expected however when I hook up to the event socket and request "event talk notalk" nothing is ever fired, any thoughts on where I am going wrong appreciated. Thanks Steve Steven Brown email steve at justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated. From brian at freeswitch.org Thu Jun 18 06:16:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 08:16:58 -0500 Subject: [Freeswitch-users] VAD, TALK and NOTALK events In-Reply-To: <3e6d7b0c0906180600q1b5f5b3dp4e28350de7aede98@mail.gmail.com> References: <3e6d7b0c0906180600q1b5f5b3dp4e28350de7aede98@mail.gmail.com> Message-ID: <866F2582-2968-4A33-9B1E-CCBBE6294FBA@freeswitch.org> I suspect you're going for TALK and NOTALK as the event names? its CUSTOM conference:: maintenance /b On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: > Hi, > > I have been trying to pick up TALK and NOTALK events but with no > success, I have enabled VAD for "both" in my config and the rtp is > stopping and starting as expected however when I hook up to the event > socket and request "event talk notalk" nothing is ever fired, any > thoughts on where I am going wrong appreciated. > > Thanks > > Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/e33c9a30/attachment-0002.html From brian at freeswitch.org Thu Jun 18 06:44:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 08:44:56 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> References: <979425.24268.qm@web111305.mail.gq1.yahoo.com> <0E625A5F-A88E-4069-9EAC-BB7CFC7934FD@freeswitch.org> <9dc4a1670906171156tdb13405gac4206bc5240ece@mail.gmail.com> Message-ID: <556D3D82-61EA-4ABB-ADEB-DF3B88B0F46B@freeswitch.org> If you're donating you can send it to my paypal brian at freeswitch.org, I also received the sound order for the zrtp sound files and a few odds and ends we needed. The order was 650 dollars and thus far I have only received a 50 dollar donation to help pay for it. So if you wanna pitch in on that also please let me know. I'm paying this out of my pocket. Thanks, Brian On Jun 17, 2009, at 1:56 PM, EdPimentl wrote: > I will match the 150.00 > > Best regards, > -E > CEO and Founder > Gpro.ws > http://Twitter.com/edpimentl > > http://TwebEX.com (Twitter Based Online Web Conference Platform) > http://TwitrShare.com (Send Picture and Message to Tweet Contacts) > http://TweetUp.ws (Twitter based MeetUp service) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/9a13f8d4/attachment-0002.html From j3flight at gmail.com Thu Jun 18 06:55:07 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Thu, 18 Jun 2009 08:55:07 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <24084409.post@talk.nabble.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <24084409.post@talk.nabble.com> Message-ID: <4A3A473B.5090803@gmail.com> Wow, I apologize for the duplicate posts. The mailing list didn't want to cooperate with me last night... j3flight wrote: > I haven't gone to the trouble (yet) of making this work, but I believe you > could use execute_application from the conference controls to do just about > anything with JavaScript... > > Here's a wiki page I created after building a JavaScript IVR for a > conference server... > http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR > > There are a couple functions in there for voicing user count, etc. So, I > believe you could stick those in a script by themselves and call them from > execute_application. Somehow, you would have to identify what user is > calling the script and what conference they're in. (You could possibly set > a session variable upon entering the conference, or parse all the > conferences until you find that session's UUID.) > > I don't know what else you're trying to do, but once you get one of them > working, the rest should follow a similar template. > > Post back if you make it work, I'm interested! > From bjbrashier at gmail.com Thu Jun 18 07:26:55 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 07:26:55 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A39BD80.8020706@gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> Message-ID: <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> What I did last night was to go ahead and modify mod_conference.c to include a new "count" conference control. I've got it getting to the right place, and spitting debug messages with the right data about which member and what the count is, but for some reason the text-to-speech isn't working. That's what I'll be tacking today. The only other things I really need to figure out are a toggle for whether or not the moderator leaving ends the conference (from a DTMF, I have to clear all endconfs or something), and a command to mute all participants. Once I have those, I'm sure everything else will be a laydown. I'm not opposed to other methods, but I am opposed to increased complexity. If I can do it all in C and XML, I prefer that to some C, some XML, some lua, some JS, etc. I'll take a closer look at your example when I get into the office to see if that's a more elegant solution than what I have. On Wed, Jun 17, 2009 at 9:07 PM, wrote: > FYI: I fixed the Wiki documentation for the lock/unlock feature. > > Bradley Brashier wrote: > > So I found one interesting thing so far: the "lock" caller control > > actually does function as a toggle, and, in fact, "unlock" does not do > > anything. This goes against wiki docs on mod_conference, but is > > helpful in this instance. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/f430164a/attachment-0002.html From msc at freeswitch.org Thu Jun 18 07:31:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:31:10 -0500 Subject: [Freeswitch-users] Freeswitch / Webserver In-Reply-To: <2557047.31431245309449504.JavaMail.root@zimbra.tng.de> References: <18011901.31411245309252614.JavaMail.root@zimbra.tng.de> <2557047.31431245309449504.JavaMail.root@zimbra.tng.de> Message-ID: <87f2f3b90906180731h6f52b47cw72deaaafdc53370e@mail.gmail.com> Where is the pastebin with all of your configuration files? -MC On Thu, Jun 18, 2009 at 2:17 AM, Rudolf Denert wrote: > Yes, I removed the tags but with no effect. I think the problem is that the > webserver doesn't look in the directory where featuers.xml is deposited on > the freeswitchserver (/opt/freeswitch/conf/dialplan/). > > The issue is that FS finds the context when dialplan is the directory > /opt/freeswitch/conf/dialplan/public/ . > > But when it is on the webserver (it is on another server with a different > IP-address) I get the error that I told you. > > What should i verify in the default config. > > Greetz > > ----- Urspr?ngliche Mail ----- > Von: "Brian West" > An: freeswitch-users at lists.freeswitch.org > Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 > Amsterdam/Berlin/Bern/Rom/Stockholm/Wien > Betreff: Re: [Freeswitch-users] Freeswitch / Webserver > > Its clearly telling you that context features doesn't exist... did you > remove the context tags around your extension so that it would be in > the correct context? Review the default config again. > > /b > > On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > > > Context features not found > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > -- > Rudolf Denert, Technical Support > TNG AG, NGN > Projensdorfer Str. 324, D-24106 Kiel, Germany > phone: +49 431 7097-10, fax: +49 431 7097-555 > mailto: rdenert at tng.de http://www.tng.de > - > Register: Amtsgericht Kiel, HRB 6596 KI > Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) > Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas > Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg > Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 > - > This e-mail may contain confidential and/or privileged information. If > you are not the intended recipient (or have received this e-mail in > error) please notify the sender immediately and destroy this e-mail. Any > unauthorized copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte > Informationen. Wenn Sie nicht der richtige Adressat sind oder diese > E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den > Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie > die unbefugte Weitergabe dieser Mail ist nicht gestattet. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/da739caf/attachment-0002.html From msc at freeswitch.org Thu Jun 18 07:33:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:33:29 -0500 Subject: [Freeswitch-users] Is freeswitch can call mobile phones? In-Reply-To: <24088222.post@talk.nabble.com> References: <24088222.post@talk.nabble.com> Message-ID: <87f2f3b90906180733w41698ec2pc8a5f0a3e5bc0619@mail.gmail.com> I am not aware of anyone who will give you free access to any kind of PSTN network. If you do find someone please let us in on the secret. :) -MC On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz wrote: > > Hi is there any possible free sites ip that i can connect so I can could to > any mobiles phones? > > I know some several ip sites has the capability to call for free Ip to > Voip... I know freeswitch can do this > > > Can you give me an example site? > -- > View this message in context: > http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/9de377b4/attachment-0002.html From msc at freeswitch.org Thu Jun 18 07:35:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:35:21 -0500 Subject: [Freeswitch-users] Use Freeswitch to provide a SIP Video/Voice email service In-Reply-To: <24086865.post@talk.nabble.com> References: <24086865.post@talk.nabble.com> Message-ID: <87f2f3b90906180735w4616baa0k4899bbfcbbdcb04a@mail.gmail.com> On Thu, Jun 18, 2009 at 1:03 AM, Bilbo wrote: > > Hi, > > I would like to use Freeswitch to provide a Video/Voice mail service that > is > integrated with an email service. > > I would like to have the ability to email the Video/Voice messages as well > as the SIP users being able to collect their Video messages using their > video soft-phones. > > Has anyone done this before or know if Freeswitch is capable? > I'm sure that FS has all the hooks necessary, but it's like the proverbial Lego bricks: some assembly required. If someone has done this kind of thing already then we'd love to hear about it. -MC > > Thanks > -- > View this message in context: > http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-service-tp24086865p24086865.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/7de248f6/attachment-0002.html From maxim.tsvetov at gmail.com Thu Jun 18 07:49:59 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Thu, 18 Jun 2009 18:49:59 +0400 Subject: [Freeswitch-users] CTI Message-ID: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> Hello! We are seeking possibilities to use CTI features with Freeswitch. This features are: - click-to-dial - call popup - answer call,hangup - call transfer Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..) or there is already written module or third-party software? This solution should support 100-150 simultaneous ?onnections from freeswitch users. Could you please share you experience with CTI. Regards, Maxim Tsvetov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/f0277967/attachment-0002.html From msc at freeswitch.org Thu Jun 18 07:53:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 09:53:36 -0500 Subject: [Freeswitch-users] Automatic call distribution In-Reply-To: <45f609f90906180306n2eec6ed3lb8cf84177faf2791@mail.gmail.com> References: <45f609f90906180306n2eec6ed3lb8cf84177faf2791@mail.gmail.com> Message-ID: <87f2f3b90906180753g189fcc24s572fb32e160578d6@mail.gmail.com> On Thu, Jun 18, 2009 at 5:06 AM, selva kumar wrote: > Hi, > I have setup FS for both inbound and outbound.It is working fine. > Now I would like to configure Automatic Call Distribution(ACD).How to > configure it in Freeswitch? > > Start with this: http://wiki.freeswitch.org/wiki/Mod_fifo You can set up agents to be off-hook or on-hook and they can wait for calls. Enjoy! -MC > > Sam > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/6375b60a/attachment-0002.html From msc at freeswitch.org Thu Jun 18 08:19:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 10:19:00 -0500 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! Message-ID: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/7f847a9b/attachment-0002.html From diego.viola at gmail.com Thu Jun 18 08:26:18 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 18 Jun 2009 11:26:18 -0400 Subject: [Freeswitch-users] Which GSM gateway to buy? In-Reply-To: <698401620906180126n454589a7y3bc2ff4ae7004741@mail.gmail.com> References: <86a32abc0906161439v89fbb58kcfe8297687dee600@mail.gmail.com> <24067617.post@talk.nabble.com> <5a8712120906171326r41361863v14834a4ebc2c4574@mail.gmail.com> <698401620906171431t10432015xa78976a401dc5c5b@mail.gmail.com> <5a8712120906171443v40ee8a35n4b6cdb34a1a0acf5@mail.gmail.com> <698401620906180126n454589a7y3bc2ff4ae7004741@mail.gmail.com> Message-ID: <86a32abc0906180826y3c81599ese2279ba68810a3c@mail.gmail.com> Thanks for the suggestions guys, I think I will go with PORTech for now. @Jo?o Mesquita: Let me know when mod_khomp is done, I might consider getting some khomps in the future when the module is ready. Regards, Diego On Thu, Jun 18, 2009 at 4:26 AM, Jan Kubr wrote: > There are gateways that allow you to set your own caller ID? I thought it'd > always use the number of the SIM. > Jan > > On Thu, Jun 18, 2009 at 12:28 AM, jay binks wrote: >> >> Ive used these in the past. >> >> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >> >> sound fine, work well... >> reliable etc etc.. >> >> things to watch out for...? : >> >> *? cant send your own caller ID from them ( in my experience its locked to >> the sim ) >> *? your provider might block the IMEI number of the GSM terminal, if they >> dont like what your doing. >> >> just some stuff to consider. >> >> >> Jay >> >> >> >> >> 2009/6/18 Jo?o Mesquita >>> >>> Pricewise, is it worth it? >>> >>> jmesquita >>> >>> On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr wrote: >>>> >>>> We plan to buy one of these: >>>> >>>> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html >>>> since you can use SMTP/POP3 to manage SMS. >>>> Jan >>>> 2009/6/17 Jo?o Mesquita >>>>> >>>>> Guys, I was looking at the advantages and disadvantages of having a GSM >>>>> gateway vs. a GSM board. >>>>> >>>>> The conclusions I get are: >>>>> >>>>> Board pros >>>>> >>>>> 1. Boards are able to get/send SMS without SIP "tricks" >>>>> 2. You don't have to make a SIP call to check if channel is available >>>>> and don't rely o SIP messages to get channel status >>>>> 3. FS will be able to check for signal level on the board and fire >>>>> events on pre-defined thresholds. >>>>> >>>>> Gateway pros >>>>> >>>>> 1. I think of is the a GW can be used by more then one server, >>>>> therefore, can have failover. >>>>> 2. A GW is more scalable >>>>> >>>>> It would be nice if you, that have already used GSM GWs in production, >>>>> could comment on this. >>>>> >>>>> Thanks, >>>>> >>>>> jmesquita >>>>> >>>>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> look at www.kuhnt.com. It?s a german page. There you can find >>>>>> "Kontakt" >>>>>> where you can ask for special requirements. >>>>>> >>>>>> NOx >>>>>> >>>>>> >>>>>> >>>>>> Diego Viola wrote: >>>>>> > >>>>>> > Hi everyone, >>>>>> > >>>>>> > Can you please recommend me some GSM gateway? I'm currently looking >>>>>> > for a good one to buy... anyone have experience PORTech GSM >>>>>> > gateways? >>>>>> > Are they good? >>>>>> > >>>>>> > I also need it to work with FS, I'm also kinda new with VoIP >>>>>> > hardware. >>>>>> > >>>>>> > Thanks, >>>>>> > >>>>>> > Diego >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Freeswitch-users mailing list >>>>>> > Freeswitch-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > >>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html >>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Sincerely >> >> Jay >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve at justfone.com Thu Jun 18 08:34:37 2009 From: steve at justfone.com (Steven Brown) Date: Thu, 18 Jun 2009 16:34:37 +0100 Subject: [Freeswitch-users] VAD, TALK and NOTALK events (Brian West) Message-ID: <3e6d7b0c0906180834r72dc14adre4e6b3ae25460576@mail.gmail.com> Thanks Brian, Yes I had been looking for TALK and NOTALK, CUSTOM conference::maintenance works great. Steve > > > Message: 4 > Date: Thu, 18 Jun 2009 08:16:58 -0500 > From: Brian West > Subject: Re: [Freeswitch-users] VAD, TALK and NOTALK events > To: freeswitch-users at lists.freeswitch.org > Message-ID: <866F2582-2968-4A33-9B1E-CCBBE6294FBA at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > I suspect you're going for TALK and NOTALK as the event names? > > its CUSTOM conference:: maintenance > > /b > > > > On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: > > > Hi, > > > > I have been trying to pick up TALK and NOTALK events but with no > > success, I have enabled VAD for "both" in my config and the rtp is > > stopping and starting as expected however when I hook up to the event > > socket and request "event talk notalk" nothing is ever fired, any > > thoughts on where I am going wrong appreciated. > > > > Thanks > > > > Steve > > > > - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/a715869a/attachment-0002.html From toofics at gmail.com Thu Jun 18 09:20:05 2009 From: toofics at gmail.com (Victor Toofic) Date: Thu, 18 Jun 2009 11:20:05 -0500 Subject: [Freeswitch-users] call quality problems in conference Message-ID: <1245342005.13905.39.camel@ktulu> Hi all! I'm having some troubles with call quality using conferences. The scenario is like this: An agent makes a call to freeswitch and enters in a conference room waiting for outbound calls; on the other side there is an application generating outbound calls and when one is answered it is assigned to the first agent available, so the outbound call enters in some of the agents's conference room (it is some kind of semi-predictive dialer). I'm using conferences because we need special features like monitoring or whispering to the agents. There are times when some of the outbound calls that enter in a conference room have really bad quality: broken/choppy voice, echo, etc, Something like this: http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav This occurs in 20%-40% of the outbound calls. I know it might be because of the jitter or packet loss with our voip provider. But.. this hardly occurs when the agents dial manually (using the bridge app); when dialing manually the problem (when it ocurrs) is always unperceptible. Thats why I think the conference room is aggravating the problem. Im using the 'jitterbuffer_msec=180' in the originate command and the same in the dialplan (when the agents log-in). What do you think is happening here? Am I missing something? Any guidance will be really appreciated! Thnks!! -- Regards.. Victor Toofic From brian at freeswitch.org Thu Jun 18 09:29:34 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 11:29:34 -0500 Subject: [Freeswitch-users] call quality problems in conference In-Reply-To: <1245342005.13905.39.camel@ktulu> References: <1245342005.13905.39.camel@ktulu> Message-ID: Please post bugs to http://jira.freeswitch.org /b On Jun 18, 2009, at 11:20 AM, Victor Toofic wrote: > Hi all! > > I'm having some troubles with call quality using conferences. The > scenario is like this: > > An agent makes a call to freeswitch and enters in a conference room > waiting for outbound calls; on the other side there is an application > generating outbound calls and when one is answered it is assigned to > the > first agent available, so the outbound call enters in some of the > agents's conference room (it is some kind of semi-predictive dialer). > > I'm using conferences because we need special features like monitoring > or whispering to the agents. > > There are times when some of the outbound calls that enter in a > conference room have really bad quality: broken/choppy voice, echo, > etc, > Something like this: > > http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav > > This occurs in 20%-40% of the outbound calls. I know it might be > because > of the jitter or packet loss with our voip provider. > > But.. this hardly occurs when the agents dial manually (using the > bridge > app); when dialing manually the problem (when it ocurrs) is always > unperceptible. Thats why I think the conference room is aggravating > the > problem. > > Im using the 'jitterbuffer_msec=180' in the originate command and the > same in the dialplan (when the agents log-in). > > What do you think is happening here? > Am I missing something? Any guidance will be really appreciated! > > Thnks!! > > -- > Regards.. > Victor Toofic > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saeedahmad1981 at gmail.com Thu Jun 18 09:40:32 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Thu, 18 Jun 2009 18:40:32 +0200 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! In-Reply-To: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> References: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> Message-ID: Done :) Guten Appetit On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins wrote: > Hello FreeSWITCHers out there! I have it on good authority that the > FreeSWITCH developers have all convened in an undisclosed location. Rumors > that they are plotting to take over the world are not yet confirmed but I > will keep you updated as information becomes available. :) > > It would be great for all of us to show our support and appreciation to the > guys for all the hard work they've done. How many of us have had a question > answered on the IRC channel or here on the list by one of the guys? How many > of us use FreeSWITCH every day for work? If you've benefited from their hard > work then please give a little. If we can get everyone to hop on the paypal > link (on http://www.freeswitch.org) and donate $5 or $10 then we can > easily pay for a nice dinner for the guys. > > Please hit the link and let me know (off list) when you've donated. Let's > do this, people! > > -Michael > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/0c9e5c85/attachment-0002.html From msc at freeswitch.org Thu Jun 18 09:48:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 11:48:16 -0500 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! In-Reply-To: References: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> Message-ID: <87f2f3b90906180948i25031b67ga5668c4180b22820@mail.gmail.com> Thank you so much! The devs are really loving this. -MC On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad wrote: > Done :) > Guten Appetit > > On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins wrote: > >> Hello FreeSWITCHers out there! I have it on good authority that the >> FreeSWITCH developers have all convened in an undisclosed location. Rumors >> that they are plotting to take over the world are not yet confirmed but I >> will keep you updated as information becomes available. :) >> >> It would be great for all of us to show our support and appreciation to >> the guys for all the hard work they've done. How many of us have had a >> question answered on the IRC channel or here on the list by one of the guys? >> How many of us use FreeSWITCH every day for work? If you've benefited from >> their hard work then please give a little. If we can get everyone to hop on >> the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then >> we can easily pay for a nice dinner for the guys. >> >> Please hit the link and let me know (off list) when you've donated. Let's >> do this, people! >> >> -Michael >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/72e7641d/attachment-0002.html From andy at fabulous4.co.uk Thu Jun 18 09:54:30 2009 From: andy at fabulous4.co.uk (Andy) Date: Thu, 18 Jun 2009 17:54:30 +0100 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> Message-ID: <0DBFD71825C84229A9118D2D11A537D4@D810> Hi All, I've tested this new variable and everything works grand. I've tested recording to wav,mp3 and shoutcast and in all cases the sample rate is set correctly. I was about to post an entry on the wiki but I discovered a very similar variable already there called record_rate. I've tested this and it doesn't appear to work. Would you like me to replace this entry with details of the new one that does seem to work? A few more questions: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? Many thanks for sorting this one for me and for all your help. regards Andy _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 June 2009 18:52 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Fri, Jun 12, 2009 at 9:26 AM, Andy wrote: Excellent, thanks Anthony, I'll give it a go. Andy, can you report back on your success with this variable? Also, we would appreciate it if you could add an entry to the wiki on the channel_variables page. Let me know if you have any questions and I'll be glad to help. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/122e9c89/attachment-0002.html From brian at freeswitch.org Thu Jun 18 10:10:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 12:10:35 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <0DBFD71825C84229A9118D2D11A537D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> <0DBFD71825C84229A9118D2D11A537D4@D810> Message-ID: <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> On Jun 18, 2009, at 11:54 AM, Andy wrote: > 1) I notice that when I change the sample rate it automatically > changes the bit rate too. I understand why this is the case but > wondered if it was just as easy to be able to control the bitrate as > well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. > 2) When I use a sample rate other than 8000 I get a warning 'Sample > rate doesn't match'. I guess this puts some extra load on the > server. If all my calls are being recorded and all at 11025 can/ > should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/697fdd1b/attachment-0002.html From andy at fabulous4.co.uk Thu Jun 18 11:01:13 2009 From: andy at fabulous4.co.uk (Andy) Date: Thu, 18 Jun 2009 19:01:13 +0100 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> Message-ID: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5ec80288/attachment-0002.html From mrene_lists at avgs.ca Thu Jun 18 11:08:05 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 18 Jun 2009 13:08:05 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). Math On 18-Jun-09, at 1:01 PM, Andy wrote: > Thanks Brian, > > So, just to calrify will the base call always be 8kHz? > > On a related note, do you happen to know the bitrate of each open > channel/live call? Is it 16 kilobits per second like the recorded > audio? I need to do some calculations on the badwidth required to > handle a certain number of concurrent calls. > > Many thanks > Andy > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: 18 June 2009 18:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Sample rate and recordFile > > > On Jun 18, 2009, at 11:54 AM, Andy wrote: > >> 1) I notice that when I change the sample rate it automatically >> changes the bit rate too. I understand why this is the case but >> wondered if it was just as easy to be able to control the bitrate >> as well as the sample rate. > > If you're talking about mod_shout, NO. You'll end up picking an > invalid bitrate and asking why it doesn't work... been there done > that... I changed it a few months back to pick the optimal bitrate > for the sample rate. > >> 2) When I use a sample rate other than 8000 I get a warning 'Sample >> rate doesn't match'. I guess this puts some extra load on the >> server. If all my calls are being recorded and all at 11025 can/ >> should I alter the sample rate of the base call to 11025? > > NO. Your phone call is running at 8kHz, Your sound file is 11025 > and they don't match, If you were to play this file into an 8k > channel without a resample it would sound a little like satan. or a > dragging tape deck. The file has to be resampled to match the > current session rate. > > /b > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/e2c5c56a/attachment-0002.html From brian at freeswitch.org Thu Jun 18 11:08:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 13:08:08 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: The call rates we support are 8, 16,32 and 48k /b On Jun 18, 2009, at 1:01 PM, Andy wrote: > Thanks Brian, > > So, just to calrify will the base call always be 8kHz? > > On a related note, do you happen to know the bitrate of each open > channel/live call? Is it 16 kilobits per second like the recorded > audio? I need to do some calculations on the badwidth required to > handle a certain number of concurrent calls. > > Many thanks > Andy > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5d2c9dcf/attachment-0002.html From brian at freeswitch.org Thu Jun 18 11:08:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 13:08:43 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <8E2E795529F4498CBBEAA99FBE1E58D4@D810> References: <137853E7923C47B7890E39796657719E@D810><1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org><191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com><87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com><0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: <3EF08E69-1C64-44F9-B5AA-54BB1687A211@freeswitch.org> look in mod_shout you'll see my calculations.. I think it has to be multiples of 16 if I recall. /b On Jun 18, 2009, at 1:01 PM, Andy wrote: > > On a related note, do you happen to know the bitrate of each open > channel/live call? Is it 16 kilobits per second like the recorded > audio? I need to do some calculations on the badwidth required to > handle a certain number of concurrent calls. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/4ed51a0d/attachment-0002.html From dave at 3c.co.uk Thu Jun 18 11:16:00 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 18 Jun 2009 21:16:00 +0300 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> <0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> Message-ID: <1245348960.21945.1.camel@dk-d820> - plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll have a bit of headroom available. --Dave > Most calls are at 8kHz. The formula for bandwidth is sampling rate * > bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). > > > Math > > On 18-Jun-09, at 1:01 PM, Andy wrote: > > > Thanks Brian, > > > > So, just to calrify will the base call always be 8kHz? > > > > On a related note, do you happen to know the bitrate of each open > > channel/live call? Is it 16 kilobits per second like the recorded > > audio? I need to do some calculations on the badwidth required to > > handle a certain number of concurrent calls. > > > > Many thanks > > Andy > > > > > > ____________________________________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Brian West > > Sent: 18 June 2009 18:11 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Sample rate and recordFile > > > > > > > > > > On Jun 18, 2009, at 11:54 AM, Andy wrote: > > > > > 1) I notice that when I change the sample rate it automatically > > > changes the bit rate too. I understand why this is the case but > > > wondered if it was just as easy to be able to control the bitrate > > > as well as the sample rate. > > > > > > If you're talking about mod_shout, NO. You'll end up picking an > > invalid bitrate and asking why it doesn't work... been there done > > that... I changed it a few months back to pick the optimal bitrate > > for the sample rate. > > > > > 2) When I use a sample rate other than 8000 I get a warning > > > 'Sample rate doesn't match'. I guess this puts some extra load on > > > the server. If all my calls are being recorded and all at 11025 > > > can/should I alter the sample rate of the base call to 11025? > > > > NO. Your phone call is running at 8kHz, Your sound file is 11025 > > and they don't match, If you were to play this file into an 8k > > channel without a resample it would sound a little like satan. or a > > dragging tape deck. The file has to be resampled to match the > > current session rate. > > > > > > /b > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From anthony.minessale at gmail.com Thu Jun 18 11:29:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 18 Jun 2009 13:29:10 -0500 Subject: [Freeswitch-users] Sample rate and recordFile In-Reply-To: <1245348960.21945.1.camel@dk-d820> References: <137853E7923C47B7890E39796657719E@D810> <1B170A67-AADA-4D1B-8204-AA48B4C04B4F@freeswitch.org> <191c3a030906120902j550b51b5qbcce339eda883fb3@mail.gmail.com> <87f2f3b90906121052r53d314adp1e35a5303ae94256@mail.gmail.com> <0DBFD71825C84229A9118D2D11A537D4@D810> <2D07AB7B-D436-4773-920D-2D696AD4BE83@freeswitch.org> <8E2E795529F4498CBBEAA99FBE1E58D4@D810> <1245348960.21945.1.camel@dk-d820> Message-ID: <191c3a030906181129u5916ad85p5502dd80a8f7c604@mail.gmail.com> or go over the limit and you'll have Max Headroom =D On Thu, Jun 18, 2009 at 1:16 PM, David Knell wrote: > - plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll > have a bit of headroom available. > > --Dave > > > Most calls are at 8kHz. The formula for bandwidth is sampling rate * > > bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way). > > > > > > Math > > > > On 18-Jun-09, at 1:01 PM, Andy wrote: > > > > > Thanks Brian, > > > > > > So, just to calrify will the base call always be 8kHz? > > > > > > On a related note, do you happen to know the bitrate of each open > > > channel/live call? Is it 16 kilobits per second like the recorded > > > audio? I need to do some calculations on the badwidth required to > > > handle a certain number of concurrent calls. > > > > > > Many thanks > > > Andy > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > > Brian West > > > Sent: 18 June 2009 18:11 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Sample rate and recordFile > > > > > > > > > > > > > > > On Jun 18, 2009, at 11:54 AM, Andy wrote: > > > > > > > 1) I notice that when I change the sample rate it automatically > > > > changes the bit rate too. I understand why this is the case but > > > > wondered if it was just as easy to be able to control the bitrate > > > > as well as the sample rate. > > > > > > > > > If you're talking about mod_shout, NO. You'll end up picking an > > > invalid bitrate and asking why it doesn't work... been there done > > > that... I changed it a few months back to pick the optimal bitrate > > > for the sample rate. > > > > > > > 2) When I use a sample rate other than 8000 I get a warning > > > > 'Sample rate doesn't match'. I guess this puts some extra load on > > > > the server. If all my calls are being recorded and all at 11025 > > > > can/should I alter the sample rate of the base call to 11025? > > > > > > NO. Your phone call is running at 8kHz, Your sound file is 11025 > > > and they don't match, If you were to play this file into an 8k > > > channel without a resample it would sound a little like satan. or a > > > dragging tape deck. The file has to be resampled to match the > > > current session rate. > > > > > > > > > /b > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/14c36ec4/attachment-0002.html From nicolas at medularis.com Thu Jun 18 11:40:09 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 18 Jun 2009 14:40:09 -0400 Subject: [Freeswitch-users] Buy The FreeSWITCH Developers Dinner! In-Reply-To: <87f2f3b90906180948i25031b67ga5668c4180b22820@mail.gmail.com> References: <87f2f3b90906180819n304133fdh38361a8855938753@mail.gmail.com> <87f2f3b90906180948i25031b67ga5668c4180b22820@mail.gmail.com> Message-ID: <1b46b4e80906181140r5f83197bo57fccbd494232278@mail.gmail.com> Thank you for all the patience and effort. You've done a great work! Have a great meal! On Thu, Jun 18, 2009 at 12:48 PM, Michael Collins wrote: > Thank you so much! The devs are really loving this. > -MC > > > On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad wrote: > >> Done :) >> Guten Appetit >> >> On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins wrote: >> >>> Hello FreeSWITCHers out there! I have it on good authority that the >>> FreeSWITCH developers have all convened in an undisclosed location. Rumors >>> that they are plotting to take over the world are not yet confirmed but I >>> will keep you updated as information becomes available. :) >>> >>> It would be great for all of us to show our support and appreciation to >>> the guys for all the hard work they've done. How many of us have had a >>> question answered on the IRC channel or here on the list by one of the guys? >>> How many of us use FreeSWITCH every day for work? If you've benefited from >>> their hard work then please give a little. If we can get everyone to hop on >>> the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then >>> we can easily pay for a nice dinner for the guys. >>> >>> Please hit the link and let me know (off list) when you've donated. Let's >>> do this, people! >>> >>> -Michael >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/429a3e15/attachment-0002.html From mariouzae at gmail.com Thu Jun 18 09:58:38 2009 From: mariouzae at gmail.com (Mario Guerra Uzae da Silva -) Date: Thu, 18 Jun 2009 13:58:38 -0300 Subject: [Freeswitch-users] doubt of configuration Message-ID: Hi, I am new user of Freeswitch, I am having trouble doing basic configurations. Somebody could help me how to configure a simple extension? Thanks sorry for my bad english -- Mario Uzae -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/1980119a/attachment-0002.html From diego.viola at gmail.com Thu Jun 18 12:21:33 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 18 Jun 2009 15:21:33 -0400 Subject: [Freeswitch-users] doubt of configuration In-Reply-To: References: Message-ID: <86a32abc0906181221t23ffc44aid8d2804aec7cd06f@mail.gmail.com> Sure, but you need to provide more details, what do you want to do exactly? On Thu, Jun 18, 2009 at 12:58 PM, Mario Guerra Uzae da Silva - < mariouzae at gmail.com> wrote: > Hi, I am new user of Freeswitch, I am having trouble doing basic > configurations. Somebody could help me how to configure a simple extension? > > Thanks > > sorry for my bad english > > -- > Mario Uzae > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/06f7cda2/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Jun 18 12:54:47 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 18 Jun 2009 20:54:47 +0100 Subject: [Freeswitch-users] high cpu utilization Message-ID: Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/77f89043/attachment-0002.html From larclap at yahoo.com Thu Jun 18 12:54:53 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 18 Jun 2009 12:54:53 -0700 Subject: [Freeswitch-users] Some channel variables not written to cdr-csv? Message-ID: <016701c9f04e$a5f0b220$f1d21660$@com> I have defined the following template in autoload_config/cdr_csv.conf.xml: The resultant Master.csv in logs/1000.csv: "+19495551212","+19495551212","1000","default","2009-06-18 09:59:59","2009-06-18 10:00:06","2009-06-18 10:01:16","77","70","NORMAL_CLEARING","7551138e-5c29-11de-80e6-1b59605a543b" ,"75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+19 495551212 at 66.53.188.187","sofia/internal/sip:1001 at 192.168.10.101","" Both ${direction} and ${accountcode} do not have any data in the cdr file. Am I using the wrong variable names? I do see Caller-Direction with a valid value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx says that both these variables exist. Thanks for any help, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/5671fee3/attachment-0002.html From tntknight at gmail.com Thu Jun 18 13:30:40 2009 From: tntknight at gmail.com (Anthony Knight) Date: Thu, 18 Jun 2009 16:30:40 -0400 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: Message-ID: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Is this possibly an issue to do with a newer tickless kernel? see http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td23248559.html Tony On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > This one has me a little baffled. If have a recent build (in the last > week) of FS installed on two near identical HP servers. One happily runs > 400 concurrent calls at around 50% CPU. The other can only run around 50 > calls without the CPU going to 98%. Identical configs and lua script. > > > > Only diff is that the server having problems is running latest centos > 64bit, where the other is 32bit. Any suggestions of where I might start > looking? > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/46a6953a/attachment-0002.html From kristian.kielhofner at gmail.com Thu Jun 18 13:36:11 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 18 Jun 2009 16:36:11 -0400 Subject: [Freeswitch-users] Bootable ISO Daily FreeSWITCH build Message-ID: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> Hello everyone, I've setup one of my build servers to do a fresh check out of SVN trunk and build AstLinux with it every day at 2AM EST. The ISO and build log (for the curious) are available here: http://mirror.astlinux.org/freeswitch/daily/ I just ran a test build but daily builds will begin showing up this evening/morning at 2AM. I plan on keeping about 30 days worth of ISO images. They should be bootable on just about anything including VMware and various other virtualization platforms. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From bjbrashier at gmail.com Thu Jun 18 14:23:10 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 14:23:10 -0700 Subject: [Freeswitch-users] Voice lag in conference In-Reply-To: <7bcfdd290906161458n42e9479are572b462387f8adb@mail.gmail.com> References: <7bcfdd290906161051m26cb85fdm94214702f66126e7@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4602@mailman.wabash.local> <7bcfdd290906161402h4ee278edof59e3c800ff4f8d6@mail.gmail.com> <9889E3226F1C704D92C4A5708A2D668902024D4660@mailman.wabash.local> <191c3a030906161455t20949e7eg129509b9c74c1cd2@mail.gmail.com> <7bcfdd290906161458n42e9479are572b462387f8adb@mail.gmail.com> Message-ID: <7bcfdd290906181423xc474158h39268280cce20f5a@mail.gmail.com> So I rebooted, installed some OS updates, synched up, and am running again. I've also been doing closer comparisons between the conference I'm running and the same phones through VOIP to other locations (like between the phones without the conference). The lag isn't as bad as it was, a significant portion is due to the VOIP connection we've got (ie. conference aside), and yet certain phones still have more trouble through the conference than not, to the tune of at least 400ms more than the others. At this point, I'm prepared to punt -- blame the specific phones for now, and look at it again in a month or so when the project is closer to "done". But if anyone has any ideas on why certain phones would behave worse than others (a Polycom SoundPoint IP 320 SIP phone is the worst) I'm all ears. BB On Tue, Jun 16, 2009 at 2:58 PM, Bradley Brashier wrote: > Will do, just haven't had the time, yet! > > > On Tue, Jun 16, 2009 at 2:55 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> don't forget to read my suggestion too from earlier today =D >> >> >> >> On Tue, Jun 16, 2009 at 4:30 PM, Josh Moon wrote: >> >>> I was able to reduce it considerably. I can?t say it is completely >>> gone but I am very confident the ~.5 second delay I hear is because of the >>> time it takes my voice to go through the leaps and bounds of the phone >>> company to our server. I had at least a 3-5 second delay before I >>> experimented with the conference settings. >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >>> Brashier >>> *Sent:* Tuesday, June 16, 2009 5:02 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* Re: [Freeswitch-users] Voice lag in conference >>> >>> >>> >>> I'm not sure I've got the opportunity to do that at the moment, but I do >>> appreciate the point of view of a fellow product user. Were you able to >>> eliminate noticeable lag, or just reduce it to reasonable levels? >>> >>> >>> >>> I'll try to do something similar when I update to the newest trunk as >>> Anthony suggested. My copy is only a week old, but I'll try whatever has a >>> chance of working, and I know you guys have been working on conferencing >>> (the Moderator function couldn't have been timed better for me!). >>> >>> On Tue, Jun 16, 2009 at 12:01 PM, Josh Moon >>> wrote: >>> >>> I am not as knowledgeable as the developers that will respond to your >>> question but I had the same problem as you. Here is what I did to combat >>> the delay: >>> >>> >>> >>> First off I started everything from scratch. I reinstalled Linux and >>> then I reinstalled FreeSWITCH by creating .deb packages. >>> >>> I then created my own conference profile and set the sample rate to 4000 >>> and changed the energy level to 20. >>> >>> I also made sure to test the conference room from phones that were in >>> completely different areas so there wasn?t a chance for feedback or really >>> bad echoing problems. >>> >>> >>> >>> Once I knew the delay was solved I raised the sample rate to 8000. I >>> tested it to make sure it would work properly. >>> >>> >>> >>> As Michael stated, this could be your network infrastructure but I just >>> wanted to let another FreeSWITCH user know what I did to try and stop the >>> voice delay. >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bradley >>> Brashier >>> *Sent:* Tuesday, June 16, 2009 1:52 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] Voice lag in conference >>> >>> >>> >>> I'm creating a conferencing product for use in a system with >>> theoretically several hundred concurrent calls. I'm using FreeSwitch to >>> create this product, but am not only new to FreeSwitch, but also the entire >>> telecom industry as well as Open Source projects in general (I'm a >>> recovering BIOS guy). >>> >>> I've got a bare-bones conference up and running on the server, including >>> a handshake and a couple of features, and am using the default packages from >>> the current trunk, but I've noticed that voice lag is a pretty big issue. >>> Common lag times are several hundred milliseconds, and I've heard as long as >>> a second. It seems to be at least marginally specific to individual phones >>> -- certain phones have longer lag than others even on the same call. >>> >>> My question is really about what my options are. Is this just a part of >>> SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim >>> down that will help? Is this a common issue? If it's common, is it expected >>> by the marketplace? >>> >>> This message contains confidential information and is intended only for >>> the individual named. If you are not the named addressee you should not >>> disseminate, distribute or copy this e-mail. Please notify the sender >>> immediately by e-mail if you have received this e-mail by mistake and delete >>> this e-mail from your system. E-mail transmission cannot be guaranteed to be >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, arrive late or incomplete, or contain viruses. The sender >>> therefore does not accept liability for any errors or omissions in the >>> contents of this message, which arise as a result of e-mail transmission. If >>> verification is required please request a hard-copy version. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> This message contains confidential information and is intended only for >>> the individual named. If you are not the named addressee you should not >>> disseminate, distribute or copy this e-mail. Please notify the sender >>> immediately by e-mail if you have received this e-mail by mistake and delete >>> this e-mail from your system. E-mail transmission cannot be guaranteed to be >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, arrive late or incomplete, or contain viruses. The sender >>> therefore does not accept liability for any errors or omissions in the >>> contents of this message, which arise as a result of e-mail transmission. If >>> verification is required please request a hard-copy version. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/b300fb3e/attachment-0002.html From bjbrashier at gmail.com Thu Jun 18 15:24:20 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 15:24:20 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> Message-ID: <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> OK, so I did some more experimenting today. I found a problem with the code I'm using (again, this is off the current trunk, but with some small modifications): conference_member_say in mod_conference.c is simply not working. There are several messages in there that can theoretically tell the user something, but all of them are bypassed in the vanilla build because the default profile plays a wav instead of generating them on the fly. If you take out the wav, the message is supposed to be generated. So I took out the wavs, but I'm not hearing any messages. BTW, something I discovered last week: straight out-of-the-box with no other modifications, if you make any changes to the set of "default" caller controls in conference.conf.xml, they don't get taken. The "default" caller controls appear to get overwritten in a hard-coded fashion in mod_conference.c. A "feature", perhaps, but very confusing for us new users. Can we add some documentation in there to that effect, perhaps? BB On Thu, Jun 18, 2009 at 7:26 AM, Bradley Brashier wrote: > What I did last night was to go ahead and modify mod_conference.c to > include a new "count" conference control. I've got it getting to the right > place, and spitting debug messages with the right data about which member > and what the count is, but for some reason the text-to-speech isn't working. > That's what I'll be tacking today. > > The only other things I really need to figure out are a toggle for whether > or not the moderator leaving ends the conference (from a DTMF, I have to > clear all endconfs or something), and a command to mute all participants. > Once I have those, I'm sure everything else will be a laydown. > > I'm not opposed to other methods, but I am opposed to increased complexity. > If I can do it all in C and XML, I prefer that to some C, some XML, some > lua, some JS, etc. I'll take a closer look at your example when I get into > the office to see if that's a more elegant solution than what I have. > > On Wed, Jun 17, 2009 at 9:07 PM, wrote: > >> FYI: I fixed the Wiki documentation for the lock/unlock feature. >> >> Bradley Brashier wrote: >> > So I found one interesting thing so far: the "lock" caller control >> > actually does function as a toggle, and, in fact, "unlock" does not do >> > anything. This goes against wiki docs on mod_conference, but is >> > helpful in this instance. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/73e471eb/attachment-0002.html From evilla at chipoly.com Thu Jun 18 15:41:52 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Thu, 18 Jun 2009 16:41:52 -0600 Subject: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ? Message-ID: <00ca01c9f065$fa3af220$eeb0d660$@com> Hello, I am planning to build a plataform to sell content, pictures, tones, MMS, etc. Do you know wich GSM 3G boards should work? Anyone has done this? Greetings! Edwin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/50300801/attachment-0002.html From d at unwire.it Thu Jun 18 16:26:29 2009 From: d at unwire.it (Darin Weeks) Date: Thu, 18 Jun 2009 16:26:29 -0700 Subject: [Freeswitch-users] Bootable ISO Daily FreeSWITCH build In-Reply-To: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> References: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> Message-ID: <989132e70906181626n117aea5fl84e987b337374d1a@mail.gmail.com> Thanks! I added a link from the wiki... http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux On Thu, Jun 18, 2009 at 1:36 PM, Kristian Kielhofner wrote: > Hello everyone, > > ?I've setup one of my build servers to do a fresh check out of SVN > trunk and build AstLinux with it every day at 2AM EST. ?The ISO and > build log (for the curious) are available here: > > http://mirror.astlinux.org/freeswitch/daily/ > > ?I just ran a test build but daily builds will begin showing up this > evening/morning at 2AM. > > ?I plan on keeping about 30 days worth of ISO images. ?They should be > bootable on just about anything including VMware and various other > virtualization platforms. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it From msc at freeswitch.org Thu Jun 18 16:35:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 18:35:46 -0500 Subject: [Freeswitch-users] Bootable ISO Daily FreeSWITCH build In-Reply-To: <989132e70906181626n117aea5fl84e987b337374d1a@mail.gmail.com> References: <2d9149cd0906181336o6d8409bel3956009ea628d24a@mail.gmail.com> <989132e70906181626n117aea5fl84e987b337374d1a@mail.gmail.com> Message-ID: <87f2f3b90906181635l1c68c507q86574178c372f4a5@mail.gmail.com> Thanks for all of your help! On Thu, Jun 18, 2009 at 6:26 PM, Darin Weeks wrote: > Thanks! I added a link from the wiki... > http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Linux > > On Thu, Jun 18, 2009 at 1:36 PM, Kristian > Kielhofner wrote: > > Hello everyone, > > > > I've setup one of my build servers to do a fresh check out of SVN > > trunk and build AstLinux with it every day at 2AM EST. The ISO and > > build log (for the curious) are available here: > > > > http://mirror.astlinux.org/freeswitch/daily/ > > > > I just ran a test build but daily builds will begin showing up this > > evening/morning at 2AM. > > > > I plan on keeping about 30 days worth of ISO images. They should be > > bootable on just about anything including VMware and various other > > virtualization platforms. > > > > -- > > Kristian Kielhofner > > http://www.astlinux.org > > http://blog.krisk.org > > http://www.star2star.com > > http://www.submityoursip.com > > http://www.voalte.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/a47b3b51/attachment-0002.html From msc at freeswitch.org Thu Jun 18 16:38:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 18:38:19 -0500 Subject: [Freeswitch-users] Some channel variables not written to cdr-csv? In-Reply-To: <016701c9f04e$a5f0b220$f1d21660$@com> References: <016701c9f04e$a5f0b220$f1d21660$@com> Message-ID: <87f2f3b90906181638v47713ab0w490b8ba2f3f0fd00@mail.gmail.com> Do you have any way to ensure that those variables are populated? Can you manually set those in the dialplan? Also, are you doing a leg only or b leg only or both? -MC On Thu, Jun 18, 2009 at 2:54 PM, Lars Zeb wrote: > I have defined the following template in > autoload_config/cdr_csv.conf.xml: > > > > > > > > > > The resultant Master.csv in logs/1000.csv: > > > > "+19495551212","+19495551212","1000","default","2009-06-18 > 09:59:59","2009-06-18 10:00:06","2009-06-18 > 10:01:16","77","70","NORMAL_CLEARING","7551138e-5c29-11de-80e6-1b59605a543b","75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+ > 19495551212 at 66.53.188.187","sofia/internal/sip:1001 at 192.168.10.101 > ","" > > > > Both ${direction} and ${accountcode} do not have any data in the cdr file. > Am I using the wrong variable names? I do see Caller-Direction with a valid > value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki > at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx says > that both these variables exist. > > > > Thanks for any help, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/96feeb8f/attachment-0002.html From juanma.v82 at gmail.com Thu Jun 18 16:48:55 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Thu, 18 Jun 2009 20:48:55 -0300 Subject: [Freeswitch-users] Can it do it? Message-ID: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Hi, I need to have the hability to negotiate the codec in a session (using proxy media or bypass media), unfortunally I've been unable to achive this due the documentation that I've found about it's vague. I've already tried using "absolute_codec_string" and everything that says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams to ignore it when the media is in "bypass media" or "proxy media". I need to configure the FS as a SBC or as a pseudo proxy (I already know that FS is not intend to do it, but in the documentation says that it can). I've also tried to manually modify the SDP using: And updating variables "switch_r_sdp" and "switch_l_dsp" but it also seams to ignore it. Here is the config: Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 What I need, is to offer to the SWITCH only the codecs defined for Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, only offer the codecs available for Endpoint2. Eventually the SWITCH will do the transcoding. So, here is my question, is there any way to achive this? (Handle the invite codecs in bypass or proxy media), if so, is there any example to follow? o can you give a tip? Thanks in advance, Regards From j3flight at gmail.com Thu Jun 18 17:01:39 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Thu, 18 Jun 2009 19:01:39 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> Message-ID: <4A3AD563.50403@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/2e1b2651/attachment-0002.html From bjbrashier at gmail.com Thu Jun 18 17:15:14 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 17:15:14 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A3AD563.50403@gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> Message-ID: <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> I was indeed looking at "announce-count", but from the code, it looks like that was designed to announce to the caller how many people were on the conference only when they were joining and the number was over a threshold specified in the profile. Not exactly what I was looking for, but it did help me find some of the right variables. And no, it didn't work, but I see now that it's most likely because conference_member_say wasn't working. I didn't think to try to define tts_engine and tts_voice, though, thinking that things like that had likely defaults. Obviously that would be an issue if not. I'll look at that next. Don't quote me on what announce-count is supposed to do, yet -- I only looked at it for long enough to tell that it wasn't what I needed. Once I have things working the way I want, I feel like I'll have enough data to be more certain of what everything does, and then I'll be happy to help you fill those out. I like your solution on the default controls. Naming them "sample" instead of "default" would do fine. Alternately, if we put a blurb in the comments above the default controls saying "these controls are hard-coded, and changes will not be taken into account. They are here as an example only", that would probably be good enough. Also, it's not clear that the DTMF commands for caller controls can be multiple digits. It might go without saying, but I didn't think about it until a little ways in, so something on the wiki might be nice. On Thu, Jun 18, 2009 at 5:01 PM, wrote: > I also saw the option for the "announce-count" conference parameter (which > i assume is what you're trying to use) and it didn't seem to work for me > either. I couldn't figure out whether I was doing something wrong or if it > was not working - that's why I implemented it in JS. Looking at the code > now, do you have tts_engine and tts_voice defined in the conference config > file. Looks like conference_member_say won't do anything without those... > > I can definitely attest to the confusion on your second point... It took > me a while to figure out the "default" conference controls as well. As long > as you name your caller-controls something else, it all works great. The > easy fix would be to modify the included conference config file so that the > sample conference controls had a different name. If someone removed them > manually, it would work as advertised. > > The wiki doc for mod_conference still needs some help too. I tried to fill > in what I knew recently by adding all the options I could find in the source > and re-arranging the page to make it easier to understand for new folks. I > had to leave a bunch of ??? in places though because I didn't know what > something did or meant... Can anyone help complete that? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/250cb78b/attachment-0002.html From msc at freeswitch.org Thu Jun 18 17:51:14 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 18 Jun 2009 19:51:14 -0500 Subject: [Freeswitch-users] Last call: buy dinner for FreeSWITCH devs Message-ID: <59B37427-3A16-41F8-8696-EDECF4F2C9A3@freeswitch.org> FYI, the devs report that they are at the restaurant! Last chance to pitch in and feed the troops. :) hit the paypal button on the main FreeSWITCH page: http://www.freeswitch.org Keep those devs happy and fed and version 1.0.4 will be here before you know it! -MC From darklion11 at yahoo.com Thu Jun 18 18:25:58 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 18 Jun 2009 18:25:58 -0700 (PDT) Subject: [Freeswitch-users] Is freeswitch can call mobile phones? In-Reply-To: <87f2f3b90906180733w41698ec2pc8a5f0a3e5bc0619@mail.gmail.com> References: <24088222.post@talk.nabble.com> <87f2f3b90906180733w41698ec2pc8a5f0a3e5bc0619@mail.gmail.com> Message-ID: <24104115.post@talk.nabble.com> I got one... But its a secret... mercutioviz wrote: > > I am not aware of anyone who will give you free access to any kind of PSTN > network. If you do find someone please let us in on the secret. :) > > -MC > > On Thu, Jun 18, 2009 at 2:58 AM, Edmar Cruz wrote: > >> >> Hi is there any possible free sites ip that i can connect so I can could >> to >> any mobiles phones? >> >> I know some several ip sites has the capability to call for free Ip to >> Voip... I know freeswitch can do this >> >> >> Can you give me an example site? >> -- >> View this message in context: >> http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24088222.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p24104115.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Jun 18 18:42:41 2009 From: dujinfang at gmail.com (seven) Date: Fri, 19 Jun 2009 09:42:41 +0800 Subject: [Freeswitch-users] Some channel variables not written to cdr-csv? In-Reply-To: <016701c9f04e$a5f0b220$f1d21660$@com> References: <016701c9f04e$a5f0b220$f1d21660$@com> Message-ID: try to run verbose_event before answer or bridge might help. On Jun 19, 2009, at 3:54 AM, Lars Zeb wrote: > I have defined the following template in autoload_config/ > cdr_csv.conf.xml: > > "${caller_id_name}","${caller_id_number}","$ > {destination_number}","${context}","${start_stamp}","${answer_stamp}", > "${end_stamp}","${duration}","${billsec}","${hangup_cause}","$ > {uuid}","${bleg_uuid}","${accountcode}","${read_codec}","$ > {write_codec}", > "${channel_name}","${bridge_channel}","${direction}" > > > The resultant Master.csv in logs/1000.csv: > > "+19495551212","+19495551212","1000","default","2009-06-18 > 09:59:59","2009-06-18 10:00:06","2009-06-18 > 10 > : > 01 > : > 16 > ","77 > ","70 > ","NORMAL_CLEARING > ","7551138e > -5c29 > -11de > -80e6 > -1b59605a543b > ","75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+19495551212 at 66.53.188.187 > ","sofia/internal/sip:1001 at 192.168.10.101","" > > Both ${direction} and ${accountcode} do not have any data in the cdr > file. Am I using the wrong variable names? I do see Caller-Direction > with a valid value ([inbound]) in freeswitch.log, but nothing like > accountcode. The wiki at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx > says that both these variables exist. > > Thanks for any help, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/616d0389/attachment-0002.html From timb0311 at hotmail.com Thu Jun 18 18:54:59 2009 From: timb0311 at hotmail.com (Tim B) Date: Thu, 18 Jun 2009 21:54:59 -0400 Subject: [Freeswitch-users] Originate fax to local extension for testing Message-ID: Trying to do a local test for faxing. Keep getting an error. Can someone tell me how to correct this? Tim default dialplan: //inbound from remote box works fine - connect asterisk box and fs box, then fax from asterisk to fs... OK - also fax from fs to asterisk.... OK // local fax on fs .... FAILS!! // my originate command: originate sofia/internal/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) // error as follows: 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing FreeSWITCH->8000 in context public 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 Legged calls 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] _________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/453d984d/attachment-0002.html From paul.degt at gmail.com Thu Jun 18 19:06:05 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Thu, 18 Jun 2009 22:06:05 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available Message-ID: <4A3AF28D.4040708@gmail.com> http://versafon.com/versafonweb/Software.jsp Essentially it's a wrapper around inbound socket interface, not all events supported yet, and not all event parameters/variables. It's multi threaded and scaled well in testing. We offer commercial support and development for FreeSwitch as well. From jmesquita at gmail.com Thu Jun 18 19:30:56 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 18 Jun 2009 23:30:56 -0300 Subject: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ? In-Reply-To: <00ca01c9f065$fa3af220$eeb0d660$@com> References: <00ca01c9f065$fa3af220$eeb0d660$@com> Message-ID: <5a8712120906181930n50d66721m648159ee4a8cbc5a@mail.gmail.com> Right now, I am working on a board that will soon support all those features but it isn't compatible to FreeSWITCH just yet. Other then that, there was thread here before discussing PorTech GSM gateways. They might be able to help. If you are interested in using other platform with the Khomp boards, I can provide you a contact. Just get in touch with me offlist. Thanks, jmesquita On Thu, Jun 18, 2009 at 7:41 PM, Ing. Edwin Villarreal wrote: > Hello, I am planning to build a plataform to sell content, pictures, > tones, MMS, etc. > > > > Do you know wich GSM 3G boards should work? Anyone has done this? > > > > *Greetings!* > > *Edwin* > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/ae4bdf28/attachment-0002.html From j3flight at gmail.com Thu Jun 18 19:38:44 2009 From: j3flight at gmail.com (j3flight) Date: Thu, 18 Jun 2009 19:38:44 -0700 (PDT) Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <4A381185.9060806@freeswitch.org> <7bcfdd290906161633t38508a90nb2ea6d14e1ef5737@mail.gmail.com> <6b65470d0906161704w793cb1ffha763c7eacf0e5b60@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> Message-ID: <24104639.post@talk.nabble.com> As far as using multiple digits in the conference controls, that doesn't seem possible. I was hoping I could make all the commands require a preceding *, like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that could be added, but then you have other silly issues to worry about... i.e. what if someone defines *1 and *10? Anyway, the conference app is powerful, especially if you want to leverage the event socket (which I have yet to try, but I can tell that's where all the goodies are). Asterisk's MeetMe has more features out of the box, but is not nearly as easily customized. I feel like mod_conference needs the following things so new folks don't go cross-eyed trying to get it to work (and I'll be more than happy to assist with this where I can): -- if the TTS stuff is required for other features to work, it needs to be turned on by default (tts is built by default now, right?) -- a great number of the possible conference parameters are missing from the default config file. I've stuck all the possibilities on the wiki (with missing descriptions in many cases) but those need to be in the default config with better explanations. (or, it could be left off the wiki entirely and a link to the default config file could be used, so documentation is only kept in one place) -- Some explanation that the "default" caller controls are HARD-CODED. I'll take a look at the wiki in just a minute and clear it up, but the config file needs an explanation too. Maybe they should be commented (or removed entirely) just to prove that you get the default set of caller controls without them being defined...?? -- View this message in context: http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24104639.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Jun 18 20:01:23 2009 From: dujinfang at gmail.com (seven) Date: Fri, 19 Jun 2009 11:01:23 +0800 Subject: [Freeswitch-users] doubt of configuration In-Reply-To: References: Message-ID: <0C8D6930-D87D-40A1-9869-C0F47C59562A@gmail.com> extensions from 1000 - 1019 are available with password 1234 by default conf On Jun 19, 2009, at 12:58 AM, Mario Guerra Uzae da Silva - wrote: > Hi, I am new user of Freeswitch, I am having trouble doing basic > configurations. Somebody could help me how to configure a simple > extension? > > Thanks > > sorry for my bad english > > -- > Mario Uzae > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Jun 18 20:17:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 18 Jun 2009 22:17:08 -0500 Subject: [Freeswitch-users] Last call: buy dinner for FreeSWITCH devs In-Reply-To: <59B37427-3A16-41F8-8696-EDECF4F2C9A3@freeswitch.org> References: <59B37427-3A16-41F8-8696-EDECF4F2C9A3@freeswitch.org> Message-ID: <48E5A4B0-5199-4366-937E-B56D8205F59E@freeswitch.org> I would like to thank everyone for Dinner... we had a great time... now MORE CODE!!! /b On Jun 18, 2009, at 7:51 PM, Michael S Collins wrote: > FYI, the devs report that they are at the restaurant! Last chance to > pitch in and feed the troops. :) hit the paypal button on the main > FreeSWITCH page: > http://www.freeswitch.org > > Keep those devs happy and fed and version 1.0.4 will be here before > you know it! > > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bjbrashier at gmail.com Thu Jun 18 20:30:00 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 20:30:00 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <24104639.post@talk.nabble.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> Message-ID: <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> I've been using multiple digits successfully right from the start, about 2 or 3 weeks ago. They do the separation of *1 and *10 the same way as several other systems -- by time. If you dial *, then 1, then wait past a timeout, then 0, you'll get *1, and *10 if you did it faster. I've tested by using 3 and 34 as separate commands, and I'm using * commands on my working system. Perhaps you should try again? Obviously, if you were confused, the docs on this could definitely be better. I'll check out the TTS stuff in the morning, and figure out those other parameters after that. Unless the original author wants to pipe up, of course. On Thu, Jun 18, 2009 at 7:38 PM, j3flight wrote: > > As far as using multiple digits in the conference controls, that doesn't > seem > possible. I was hoping I could make all the commands require a preceding > *, > like *1 for mute, *2 for lock, etc but that didn't work. I'm sure that > could be added, but then you have other silly issues to worry about... > i.e. > what if someone defines *1 and *10? > > Anyway, the conference app is powerful, especially if you want to leverage > the event socket (which I have yet to try, but I can tell that's where all > the goodies are). Asterisk's MeetMe has more features out of the box, but > is not nearly as easily customized. > > I feel like mod_conference needs the following things so new folks don't go > cross-eyed trying to get it to work (and I'll be more than happy to assist > with this where I can): > -- if the TTS stuff is required for other features to work, it needs to be > turned on by default (tts is built by default now, right?) > -- a great number of the possible conference parameters are missing from > the > default config file. I've stuck all the possibilities on the wiki (with > missing descriptions in many cases) but those need to be in the default > config with better explanations. (or, it could be left off the wiki > entirely and a link to the default config file could be used, so > documentation is only kept in one place) > -- Some explanation that the "default" caller controls are HARD-CODED. > I'll > take a look at the wiki in just a minute and clear it up, but the config > file needs an explanation too. Maybe they should be commented (or removed > entirely) just to prove that you get the default set of caller controls > without them being defined...?? > -- > View this message in context: > http://www.nabble.com/Controlling-Conference-Controls-tp24063307p24104639.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/cdbc021b/attachment-0002.html From msc at freeswitch.org Thu Jun 18 20:49:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 18 Jun 2009 22:49:17 -0500 Subject: [Freeswitch-users] Originate fax to local extension for testing In-Reply-To: References: Message-ID: <87f2f3b90906182049v6fa04d89pcb8c8bc2fe305bc2@mail.gmail.com> Tim, We need some information, specifically we need you to turn on debugging at the console and give us the log from start of call to the very end. Go to the CLI and press F8 (or type "console loglevel debug") and then initiate the call. Capture everything from the CLI from start to finish, then drop it in a pb at pastebin.freeswitch.org. Send us back the pastebin number and we'll try and diagnose it. -MC On Thu, Jun 18, 2009 at 8:54 PM, Tim B wrote: > Trying to do a local test for faxing. Keep getting an error. Can someone > tell me how to correct this? > > Tim > > > > default dialplan: > > > > > > > > > > data="last_fax=${caller_id_number}-${strftime(%Y%m%d%H%M%S)}"/> > > > > > > > > > > > > //inbound from remote box works fine > > - connect asterisk box and fs box, then fax from asterisk to fs... OK > > - also fax from fs to asterisk.... OK > > > > // local fax on fs .... FAILS!! > > // my originate command: > > originate sofia/internal/8000 at 192.168.10.35&txfax(storage/fax/test.tif) > > > > // error as follows: > > 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing > FreeSWITCH->8000 in context public > 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 > Legged calls > 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 > at 192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > > > ------------------------------ > Insert movie times and more without leaving Hotmail?. See how. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/3d16dc66/attachment-0002.html From dave at 3c.co.uk Thu Jun 18 21:31:35 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 19 Jun 2009 07:31:35 +0300 Subject: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ? In-Reply-To: <00ca01c9f065$fa3af220$eeb0d660$@com> References: <00ca01c9f065$fa3af220$eeb0d660$@com> Message-ID: <1245385895.4231.3.camel@dk-d820> Hi Edwin, Rather than using a GSM/3G card, you might do better to find a mobile services aggregator which covers the locations you're interested in - MBlox or Sybase365 would be two places to start - and use them. You'll get scalability, better reliability, etc.; be warned that MMS is *still* a pain. --Dave > Hello, I am planning to build a plataform to sell content, pictures, > tones, MMS, etc. > > > > Do you know wich GSM 3G boards should work? Anyone has done this? > > > > Greetings! > > Edwin > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From j3flight at gmail.com Thu Jun 18 21:45:41 2009 From: j3flight at gmail.com (j3flight at gmail.com) Date: Thu, 18 Jun 2009 23:45:41 -0500 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906171118l250da9edke98ce520945c7892@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> Message-ID: <4A3B17F5.9070400@gmail.com> Well crap, I must have had something else screwed up then with the multiple digits... I will try it out again soon, thanks for putting me back on track. I made some more changes to the wiki that hopefully clean up some confusion on a few things like the caller-controls. From bjbrashier at gmail.com Thu Jun 18 22:14:51 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Thu, 18 Jun 2009 22:14:51 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <4A3B17F5.9070400@gmail.com> References: <7bcfdd290906161426p74eecd0wf6a003f9c9536d78@mail.gmail.com> <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> Message-ID: <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> Actually, that's another good reason to do those wiki and/or code & comments changes... most likely, the reason you thought it couldn't be done is that you tried it and it didn't work... but you tried it on the default profile before you realized that it was hard coded. I know that's what I did and was confused about at first. I want to take a second and point out that while I may be complaining about some difficulties I'm having, the process has actually been FAR easier and faster than I had ever expected. This is a nice, solid product that works amazingly well amazingly quickly. I've been working with it for exactly 5 weeks now, starting from not knowing what SIP was or even that it had anything to do with VoIP. I've got a decent, demo conference bridge working, and am likely to be saving my company a good chunk of change as soon as I work out a few more kinks. A 2-month time to market from complete zero is just incredible. On Thu, Jun 18, 2009 at 9:45 PM, wrote: > Well crap, I must have had something else screwed up then with the > multiple digits... I will try it out again soon, thanks for putting me > back on track. I made some more changes to the wiki that hopefully > clean up some confusion on a few things like the caller-controls. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/05333b4a/attachment-0002.html From mattdfong at gmail.com Thu Jun 18 23:16:35 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 18 Jun 2009 23:16:35 -0700 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | Message-ID: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> I have two providers and want to first try to originate the call with provider A, and if that fails on certain failure causes attempt to originate the same call with provider B. Normally I would do this using an | in the dial string like originate sofia/gatewayA/123456|sofia/gatewayB/123456 but I do not want it to fail over on failure codes like USER_BUSY or NO_ANSWER because then I'm simply wasting the second carrier's resources. instead I would like to set a which error codes are considered a failure. The wiki notes a failure_causes channel variable for bridged calls, but this does not seem to work in an originate statement like originate {failure_causes='RECOVERY_ON_TIMER_EXPIRE',continue_on_fail=false}sofia/gateway/ gatewaya.com/1XXXXXX |sofia/gateway/gatewayb.com/1XXXXXX 5000 Can anyone recommend a way to accomplish what I'm trying to do...preferably w/o mod_lcr? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/20a18063/attachment-0002.html From mrene_lists at avgs.ca Thu Jun 18 23:19:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 19 Jun 2009 01:19:43 -0500 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> Message-ID: and then originate it. Math On 19-Jun-09, at 1:16 AM, Matthew Fong wrote: > I have two providers and want to first try to originate the call > with provider A, and if that fails on certain failure causes attempt > to originate the same call with provider B. > > Normally I would do this using an | in the dial string like > originate sofia/gatewayA/123456|sofia/gatewayB/123456 > > but I do not want it to fail over on failure codes like USER_BUSY or > NO_ANSWER because then I'm simply wasting the second carrier's > resources. instead I would like to set a which error codes are > considered a failure. The wiki notes a failure_causes channel > variable for bridged calls, but this does not seem to work in an > originate statement like > > originate > {failure_causes > ='RECOVERY_ON_TIMER_EXPIRE',continue_on_fail=false}sofia/gateway/ > gatewaya.com/1XXXXXX |sofia/gateway/gatewayb.com/1XXXXXX 5000 > > Can anyone recommend a way to accomplish what I'm trying to > do...preferably w/o mod_lcr? > > Thanks. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/733fe4bd/attachment-0002.html From jason at jasonjgw.net Thu Jun 18 23:31:09 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 19 Jun 2009 16:31:09 +1000 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> Message-ID: <20090619063109.GA21842@jdc.jasonjgw.net> Mathieu Rene wrote: > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then > originate it. Or if you're originating from a script, set that as a channel variable first. From mattdfong at gmail.com Thu Jun 18 23:38:58 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 18 Jun 2009 23:38:58 -0700 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <20090619063109.GA21842@jdc.jasonjgw.net> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> <20090619063109.GA21842@jdc.jasonjgw.net> Message-ID: <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> the script is not part of a session or dial plan. :( On Thu, Jun 18, 2009 at 11:31 PM, Jason White wrote: > Mathieu Rene wrote: > > > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then > > originate it. > > Or if you're originating from a script, set that as a channel variable > first. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090618/caa4443b/attachment-0002.html From jason at jasonjgw.net Thu Jun 18 23:43:25 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 19 Jun 2009 16:43:25 +1000 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A39BD80.8020706@gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> Message-ID: <20090619064325.GA22568@jdc.jasonjgw.net> Bradley Brashier wrote: > I want to take a second and point out that while I may be complaining about > some difficulties I'm having, the process has actually been FAR easier and > faster than I had ever expected. This is a nice, solid product that works > amazingly well amazingly quickly. I've been working with it for exactly 5 > weeks now, starting from not knowing what SIP was or even that it had > anything to do with VoIP. I've got a decent, demo conference bridge working, > and am likely to be saving my company a good chunk of change as soon as I > work out a few more kinks. A 2-month time to market from complete zero is > just incredible. If organizations such as yours could contribute some of the money they save to FreeSWITCH development, for example to get desired features implemented or bugs fixed, I'm sure that would help to make what is already a great project even better. The software is excellent, the community helpful, and I've had a lot of fun experimenting with IPv6, TLS, ZRTP and other features that few other projects have implemented. From nik.middleton at noblesolutions.co.uk Fri Jun 19 01:23:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 19 Jun 2009 09:23:20 +0100 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I don't think that's the issue Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Knight Sent: 18 June 2009 21:31 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] high cpu utilization Is this possibly an issue to do with a newer tickless kernel? see http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td2324855 9.html Tony On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton wrote: Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/be51d9a6/attachment-0002.html From timb0311 at hotmail.com Fri Jun 19 02:31:08 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 05:31:08 -0400 Subject: [Freeswitch-users] Originate fax to local extension for testing In-Reply-To: References: Message-ID: Michael, I ran the debugging you asked. I also tried to post it to pastebin.freeswitch.org but can't login. I used my login for the freeswitch site, but that doesn't seem to work?? How do I gain acess? Thanks. Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 18 Jun 2009 22:49:17 -0500 > From: Michael Collins > Subject: Re: [Freeswitch-users] Originate fax to local extension for > testing > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906182049v6fa04d89pcb8c8bc2fe305bc2 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Tim, > > We need some information, specifically we need you to turn on debugging at > the console and give us the log from start of call to the very end. Go to > the CLI and press F8 (or type "console loglevel debug") and then initiate > the call. Capture everything from the CLI from start to finish, then drop it > in a pb at pastebin.freeswitch.org. Send us back the pastebin number and > we'll try and diagnose it. > > -MC > _________________________________________________________________ Hotmail? has ever-growing storage! Don?t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/df43d281/attachment-0002.html From jason at jasonjgw.net Fri Jun 19 02:43:46 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 19 Jun 2009 19:43:46 +1000 Subject: [Freeswitch-users] Originate fax to local extension for testing In-Reply-To: References: Message-ID: <20090619094346.GA8755@jdc.jasonjgw.net> Tim B wrote: > > Michael, I ran the debugging you asked. I also tried to post it to > pastebin.freeswitch.org but can't login. I used my login for the freeswitch > site, but that doesn't seem to work?? How do I gain acess? When I connect to pastebin.freeswitch.org I get a helpful notice saying the login and password is pastebin/freeswitch From darklion11 at yahoo.com Fri Jun 19 04:12:55 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 19 Jun 2009 04:12:55 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? Message-ID: <24109532.post@talk.nabble.com> My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/1006 at 116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1006 at 116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1006 at 116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 I put this on my external and internal profile And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24109532.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Fri Jun 19 04:13:30 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 19 Jun 2009 04:13:30 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? Message-ID: <24109532.post@talk.nabble.com> My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/1006 at 116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1006 at 116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1006 at 116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 I put this on my external and internal profile param name="apply-inbound-acl" value="globals"/> And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24109532.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From saeedahmad1981 at gmail.com Fri Jun 19 05:06:46 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 19 Jun 2009 14:06:46 +0200 Subject: [Freeswitch-users] Can it do it? In-Reply-To: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Message-ID: It seems that you want to transcode G729 which is not possible. You can use it in passthru mode only. On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote: > Hi, > > I need to have the hability to negotiate the codec in a session (using > proxy media or bypass media), unfortunally I've been unable to achive > this due the documentation that I've found about it's vague. > > I've already tried using "absolute_codec_string" and everything that > says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams > to ignore it when the media is in "bypass media" or "proxy media". I > need to configure the FS as a SBC or as a pseudo proxy (I already know > that FS is not intend to do it, but in the documentation says that it > can). > > I've also tried to manually modify the SDP using: > > > And updating variables "switch_r_sdp" and "switch_l_dsp" but it also > seams to ignore it. > > Here is the config: > > Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 > > What I need, is to offer to the SWITCH only the codecs defined for > Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, > only offer the codecs available for Endpoint2. Eventually the SWITCH > will do the transcoding. > > So, here is my question, is there any way to achive this? (Handle the > invite codecs in bypass or proxy media), if so, is there any example > to follow? o can you give a tip? > > Thanks in advance, > Regards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/480ba462/attachment-0002.html From saeedahmad1981 at gmail.com Fri Jun 19 05:08:50 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 19 Jun 2009 14:08:50 +0200 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Message-ID: BTW, what is SWITCH? Can it do transcoding? On Fri, Jun 19, 2009 at 2:06 PM, Saeed Ahmad wrote: > It seems that you want to transcode G729 which is not possible. You can use > it in passthru mode only. > > > On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote: > >> Hi, >> >> I need to have the hability to negotiate the codec in a session (using >> proxy media or bypass media), unfortunally I've been unable to achive >> this due the documentation that I've found about it's vague. >> >> I've already tried using "absolute_codec_string" and everything that >> says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams >> to ignore it when the media is in "bypass media" or "proxy media". I >> need to configure the FS as a SBC or as a pseudo proxy (I already know >> that FS is not intend to do it, but in the documentation says that it >> can). >> >> I've also tried to manually modify the SDP using: >> >> >> And updating variables "switch_r_sdp" and "switch_l_dsp" but it also >> seams to ignore it. >> >> Here is the config: >> >> Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 >> >> What I need, is to offer to the SWITCH only the codecs defined for >> Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, >> only offer the codecs available for Endpoint2. Eventually the SWITCH >> will do the transcoding. >> >> So, here is my question, is there any way to achive this? (Handle the >> invite codecs in bypass or proxy media), if so, is there any example >> to follow? o can you give a tip? >> >> Thanks in advance, >> Regards >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/fa5d19cb/attachment-0002.html From timb0311 at hotmail.com Fri Jun 19 05:39:13 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 08:39:13 -0400 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 In-Reply-To: References: Message-ID: here is the log... http://pastebin.freeswitch.org/9440 haha, yeah i see it now... duh. pulled an all nighter, too many things going on. must have overlooked it. > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > the login and password is pastebin/freeswitch > been trying to break myself into freeswitch on top of my original workload. thanks for the help. _________________________________________________________________ Lauren found her dream laptop. Find the PC that?s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c53ba6fc/attachment-0002.html From brian at freeswitch.org Fri Jun 19 06:06:31 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Jun 2009 08:06:31 -0500 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> <20090619063109.GA21842@jdc.jasonjgw.net> <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> Message-ID: <58F248FD-6963-4171-9589-92D506DC6935@freeswitch.org> If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT issue. /b On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote: > the script is not part of a session or dial plan. :( > > On Thu, Jun 18, 2009 at 11:31 PM, Jason White > wrote: > Mathieu Rene wrote: > > > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then > > originate it. > > Or if you're originating from a script, set that as a channel > variable first. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/1f7ccb7c/attachment-0002.html From anthony.minessale at gmail.com Fri Jun 19 07:51:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Jun 2009 09:51:35 -0500 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: <191c3a030906190751m9e04eb4lcff5463534863372@mail.gmail.com> use top -H to get the per-thread cpu usage. see if any one thread is using more than the rest. then get a gcore of the running process and do a thread apply all bt and get a bt from the thread with the matching id. Maybe that will tell you what is doing all the work. On Fri, Jun 19, 2009 at 3:23 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I?m running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly > I don?t think that?s the issue > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Knight > *Sent:* 18 June 2009 21:31 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] high cpu utilization > > > > Is this possibly an issue to do with a newer tickless kernel? > > > > see > http://www.nabble.com/FreeSWITCH-under-the-Linux-2.6.29-kernel-td23248559.html > > > > Tony > > On Thu, Jun 18, 2009 at 3:54 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi Guys, > > > > This one has me a little baffled. If have a recent build (in the last > week) of FS installed on two near identical HP servers. One happily runs > 400 concurrent calls at around 50% CPU. The other can only run around 50 > calls without the CPU going to 98%. Identical configs and lua script. > > > > Only diff is that the server having problems is running latest centos > 64bit, where the other is 32bit. Any suggestions of where I might start > looking? > > > > Regards, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/3586f84c/attachment-0002.html From msc at freeswitch.org Fri Jun 19 08:00:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 10:00:35 -0500 Subject: [Freeswitch-users] Update - Transmit fax locally for test Message-ID: <87f2f3b90906190800u5d9436cbu2bd594bc8d09503@mail.gmail.com> Tim, Look at lines 47 and 48 of the pastebin. I think something goofy is happening there. What is 8000 at x.x.x.x in your system? Is that the receive fax extension? -MC ---------- Forwarded message ---------- From: Tim B Date: Fri, Jun 19, 2009 at 7:39 AM Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 To: freeswitch-users at lists.freeswitch.org here is the log... http://pastebin.freeswitch.org/9440 haha, yeah i see it now... duh. pulled an all nighter, too many things going on. must have overlooked it. > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > the login and password is pastebin/freeswitch > been trying to break myself into freeswitch on top of my original workload. thanks for the help. ------------------------------ Lauren found her dream laptop. Find the PC that?s right for you. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c9567fad/attachment-0002.html From max.bridgewater at gmail.com Fri Jun 19 08:14:58 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 11:14:58 -0400 Subject: [Freeswitch-users] Help with Socket event again Message-ID: Any help our there? I'm still trying to get this piece working. Essentially what i wan to do is, when a call comes in (from registered devices as well as unregistered devices), notify the my server socket. Somehow it's not working. The change i made compared to the standard Freeswitch settings are the following: 1) Added following extension that in /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: 2) Changed file: /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: I noticed that with this extension, all calls received from external providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. But calls from registered devices and initiated using the socket interface are not forwarded. Is there something that need to be changed in the profiles? or is something wrong with my dial string? {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. In the logs, i cannot see that that my extension is being matched. Any idea, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/b1ac05b3/attachment-0002.html From apt.get at gmail.com Fri Jun 19 08:19:59 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 19 Jun 2009 09:19:59 -0600 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: On Fri, Jun 19, 2009 at 2:23 AM, Nik Middleton wrote: > I?m running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I > don?t think that?s the issue I could be wrong, but I think PAE is a 32-bit kernel adapted for hardware with >4GB RAM. This can create a lot of overhead compared to running a true 64-bit kernel or a 32-bit kernel without PAE. Confirm this with 'uname -a' db From msc at freeswitch.org Fri Jun 19 08:43:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 10:43:10 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: Message-ID: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> Can you turn on debugging (F8) and capture all the output after your originate? Put it into a pastebin. (pastebin.freeswitch.org) -MC On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater wrote: > Any help our there? > > I'm still trying to get this piece working. Essentially what i wan to do > is, when a call comes in (from registered devices as well as unregistered > devices), notify the my server socket. Somehow it's not working. The change > i made compared to the standard Freeswitch settings are the following: > > 1) Added following extension that in > /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: > > > > > > /> > > > > > 2) Changed file: > /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: > > > > > > > > > > > > > I noticed that with this extension, all calls received from external > providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. > But calls from registered devices and initiated using the socket interface > are not forwarded. Is there something that need to be changed in the > profiles? > > or is something wrong with my dial string? > {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. > > In the logs, i cannot see that that my extension is being matched. > > Any idea, > > Max. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/28c973c5/attachment-0002.html From bjbrashier at gmail.com Fri Jun 19 08:59:57 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 08:59:57 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <20090619064325.GA22568@jdc.jasonjgw.net> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <7bcfdd290906180726p139e519do1e394c16461536be@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> Message-ID: <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I chose flite), uncommenting a few things, and setting the TTS variables in the conference profile. You do have to rebuild FS to do this. With that working, my count function works, too. I posted a bug last night about conferencing, BTW: if you're using the wait-mod function, where the conference doesn't start until the moderator arrives, and you're also using separate profiles for users and moderators, the users only have the "user" profile until the first moderator arrives. At that time, they switch to also be using the "moderator" profile. Now that TTS is working, I'm going to see about helping you fill out those ???s, and maybe see if I can figure out how to fix the above bug. BB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c38c19e6/attachment-0002.html From mattdfong at gmail.com Fri Jun 19 10:35:37 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 10:35:37 -0700 Subject: [Freeswitch-users] Failure Causes in an Originate Statement with | In-Reply-To: <58F248FD-6963-4171-9589-92D506DC6935@freeswitch.org> References: <4256bf830906182316qff4bbc2jb4cd76d9dd447c49@mail.gmail.com> <20090619063109.GA21842@jdc.jasonjgw.net> <4256bf830906182338x7e2f48b1p9c763f7d17acc22a@mail.gmail.com> <58F248FD-6963-4171-9589-92D506DC6935@freeswitch.org> Message-ID: <4256bf830906191035x7df2b635q699d503dc96b615f@mail.gmail.com> recovery_on_timer_expire was just my example.. I actually just want to try carrier B on everything except no_answer or user_busy... On Fri, Jun 19, 2009 at 6:06 AM, Brian West wrote: > If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT issue. > /b > > On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote: > > the script is not part of a session or dial plan. :( > > On Thu, Jun 18, 2009 at 11:31 PM, Jason White wrote: > >> Mathieu Rene wrote: >> > > > data="failure_causes=user_busy,recovery_on_timer_expire" /> and then >> > originate it. >> >> Or if you're originating from a script, set that as a channel variable >> first. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/d81b1108/attachment-0002.html From max.bridgewater at gmail.com Fri Jun 19 10:58:59 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 13:58:59 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> Message-ID: Hi Mike, It's pasted here: http://pastebin.ca/1466521 Thanks, Max. On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: > Can you turn on debugging (F8) and capture all the output after your > originate? Put it into a pastebin. (pastebin.freeswitch.org) > -MC > > On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Any help our there? >> >> I'm still trying to get this piece working. Essentially what i wan to do >> is, when a call comes in (from registered devices as well as unregistered >> devices), notify the my server socket. Somehow it's not working. The change >> i made compared to the standard Freeswitch settings are the following: >> >> 1) Added following extension that in >> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >> >> >> >> >> >> > /> >> >> >> >> >> 2) Changed file: >> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >> >> >> >> >> >> >> >> >> >> >> >> >> I noticed that with this extension, all calls received from external >> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >> But calls from registered devices and initiated using the socket interface >> are not forwarded. Is there something that need to be changed in the >> profiles? >> >> or is something wrong with my dial string? >> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >> >> In the logs, i cannot see that that my extension is being matched. >> >> Any idea, >> >> Max. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/4d8f755d/attachment-0002.html From msc at freeswitch.org Fri Jun 19 11:10:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 13:10:56 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> Message-ID: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> Max, that pastebin failed miserably as none of the xml shows up. can you try again or use our pastebin.freeswitch.org site? -MC On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater wrote: > Hi Mike, > > It's pasted here: http://pastebin.ca/1466521 > > Thanks, > Max. > > > > > On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: > >> Can you turn on debugging (F8) and capture all the output after your >> originate? Put it into a pastebin. (pastebin.freeswitch.org) >> -MC >> >> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Any help our there? >>> >>> I'm still trying to get this piece working. Essentially what i wan to do >>> is, when a call comes in (from registered devices as well as unregistered >>> devices), notify the my server socket. Somehow it's not working. The change >>> i made compared to the standard Freeswitch settings are the following: >>> >>> 1) Added following extension that in >>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>> >>> >>> >>> >>> >>> >> /> >>> >>> >>> >>> >>> 2) Changed file: >>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I noticed that with this extension, all calls received from external >>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>> But calls from registered devices and initiated using the socket interface >>> are not forwarded. Is there something that need to be changed in the >>> profiles? >>> >>> or is something wrong with my dial string? >>> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >>> >>> In the logs, i cannot see that that my extension is being matched. >>> >>> Any idea, >>> >>> Max. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/49b53531/attachment-0002.html From max.bridgewater at gmail.com Fri Jun 19 11:19:09 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 14:19:09 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> Message-ID: Mike, Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to me though. Max. On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: > Max, > that pastebin failed miserably as none of the xml shows up. can you try > again or use our pastebin.freeswitch.org site? > -MC > > > On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Hi Mike, >> >> It's pasted here: http://pastebin.ca/1466521 >> >> Thanks, >> Max. >> >> >> >> >> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >> >>> Can you turn on debugging (F8) and capture all the output after your >>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>> -MC >>> >>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>> max.bridgewater at gmail.com> wrote: >>> >>>> Any help our there? >>>> >>>> I'm still trying to get this piece working. Essentially what i wan to do >>>> is, when a call comes in (from registered devices as well as unregistered >>>> devices), notify the my server socket. Somehow it's not working. The change >>>> i made compared to the standard Freeswitch settings are the following: >>>> >>>> 1) Added following extension that in >>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2) Changed file: >>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> I noticed that with this extension, all calls received from external >>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>> But calls from registered devices and initiated using the socket interface >>>> are not forwarded. Is there something that need to be changed in the >>>> profiles? >>>> >>>> or is something wrong with my dial string? >>>> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >>>> >>>> >>>> In the logs, i cannot see that that my extension is being matched. >>>> >>>> Any idea, >>>> >>>> Max. >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/f3d8db96/attachment-0002.html From bjbrashier at gmail.com Fri Jun 19 11:27:00 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 11:27:00 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <7bcfdd290906181524g473e1e82kedd6d475dca52c94@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> Message-ID: <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> So it turns out that it wasn't a bug at all -- it is a feature that was not implemented. So I've got some work to do to get that running. Since I said I would, though, here's my analysis of the conference parameters you were asking about: mute-detect-sound Different sound for if muting using only when mute-detect flag is on. max-members Specifies the maximum number of participants in a call. max-members-sound If caller cannot join because max is reached, this sound plays. Recommended if max-members is set. comfort-noise-level Sets volume of background white noise to generate. announce-count Requires TTS. When joining, tells caller how many callers are already in conference if at least the specified minimum. suppress-events ? Sets a flag, but does not appear to do anything with it. verbose-events Maximum verbosity for transcripting. timer-name Specifies the name of this profile's timer. To separate it from other timers? BB On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier wrote: > OK, I figured out the TTS stuff. It's a matter of choosing an engine (I > chose flite), uncommenting a few things, and setting the TTS variables in > the conference profile. You do have to rebuild FS to do this. > > With that working, my count function works, too. > > I posted a bug last night about conferencing, BTW: if you're using the > wait-mod function, where the conference doesn't start until the moderator > arrives, and you're also using separate profiles for users and moderators, > the users only have the "user" profile until the first moderator arrives. At > that time, they switch to also be using the "moderator" profile. > > Now that TTS is working, I'm going to see about helping you fill out those > ???s, and maybe see if I can figure out how to fix the above bug. > > BB > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/4ba96921/attachment-0002.html From bjbrashier at gmail.com Fri Jun 19 11:47:16 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 11:47:16 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> Message-ID: <7bcfdd290906191147n5742fc75mdab6525c67233500@mail.gmail.com> >mute-detect-sound >Different sound for if muting using only when mute-detect flag is on. Sorry... that didn't come out right. Try this: Different sound to play when muted by someone else and mute-detect flag is on. Plays mute-sound if this is not present. On Fri, Jun 19, 2009 at 11:27 AM, Bradley Brashier wrote: > So it turns out that it wasn't a bug at all -- it is a feature that was not > implemented. So I've got some work to do to get that running. Since I said I > would, though, here's my analysis of the conference parameters you were > asking about: > > mute-detect-sound > Different sound for if muting using only when mute-detect flag is on. > max-members > Specifies the maximum number of participants in a call. > max-members-sound > If caller cannot join because max is reached, this sound plays. Recommended > if max-members is set. > comfort-noise-level > Sets volume of background white noise to generate. > announce-count > Requires TTS. When joining, tells caller how many callers are already in > conference if at least the specified minimum. > suppress-events > ? Sets a flag, but does not appear to do anything with it. > verbose-events > Maximum verbosity for transcripting. > timer-name > Specifies the name of this profile's timer. To separate it from other > timers? > > BB > On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier wrote: > >> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I >> chose flite), uncommenting a few things, and setting the TTS variables in >> the conference profile. You do have to rebuild FS to do this. >> >> With that working, my count function works, too. >> >> I posted a bug last night about conferencing, BTW: if you're using the >> wait-mod function, where the conference doesn't start until the moderator >> arrives, and you're also using separate profiles for users and moderators, >> the users only have the "user" profile until the first moderator arrives. At >> that time, they switch to also be using the "moderator" profile. >> >> Now that TTS is working, I'm going to see about helping you fill out those >> ???s, and maybe see if I can figure out how to fix the above bug. >> >> BB >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/10c9b207/attachment-0002.html From juanma.v82 at gmail.com Fri Jun 19 12:11:58 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Fri, 19 Jun 2009 16:11:58 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> Message-ID: <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> Saeed Ahmad: Yes, it can do transcoding. Transcoding isn't the problem to my architecture, my problem is the codec negotiation between FS and Endpoints. I want to use FS as SBC (session border controller) or pseudo SIP proxy. How i said in my last e-mail FS will work in bypass-media or proxy-media in both modes FS doesn't handle media. I only want the FS handle Codec Negotiation how i described. Thanks On 19/06/2009, at 09:08, Saeed Ahmad wrote: > BTW, what is SWITCH? Can it do transcoding? > > On Fri, Jun 19, 2009 at 2:06 PM, Saeed Ahmad > wrote: > It seems that you want to transcode G729 which is not possible. You > can use it in passthru mode only. > > > On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote: > Hi, > > I need to have the hability to negotiate the codec in a session (using > proxy media or bypass media), unfortunally I've been unable to achive > this due the documentation that I've found about it's vague. > > I've already tried using "absolute_codec_string" and everything that > says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams > to ignore it when the media is in "bypass media" or "proxy media". I > need to configure the FS as a SBC or as a pseudo proxy (I already know > that FS is not intend to do it, but in the documentation says that it > can). > > I've also tried to manually modify the SDP using: > > > And updating variables "switch_r_sdp" and "switch_l_dsp" but it also > seams to ignore it. > > Here is the config: > > Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 > > What I need, is to offer to the SWITCH only the codecs defined for > Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, > only offer the codecs available for Endpoint2. Eventually the SWITCH > will do the transcoding. > > So, here is my question, is there any way to achive this? (Handle the > invite codecs in bypass or proxy media), if so, is there any example > to follow? o can you give a tip? > > Thanks in advance, > Regards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/3432d413/attachment-0002.html From mattdfong at gmail.com Fri Jun 19 12:15:37 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 12:15:37 -0700 Subject: [Freeswitch-users] CHANNEL_HANGUP event no longer having variable_billsec in header Message-ID: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> I upgraded to 13857 today, but noticed that the channel_hangup event no longer contain the variable_billsec header. Is this correct, or am I crazy? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/c7faec54/attachment-0002.html From brian at freeswitch.org Fri Jun 19 12:19:46 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Jun 2009 14:19:46 -0500 Subject: [Freeswitch-users] Can it do it? In-Reply-To: <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> Message-ID: <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> No right now you can not legally transcode G729 in FreeSWITCH, PERIOD! /b On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > Yes, it can do transcoding. Transcoding isn't the problem to my > architecture, my problem is the codec negotiation between FS and > Endpoints. > From jmesquita at gmail.com Fri Jun 19 13:05:17 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 19 Jun 2009 17:05:17 -0300 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <4A3AD563.50403@gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> Message-ID: <5a8712120906191305x5a98228eyb1847cfd2905aab5@mail.gmail.com> Inline... On Fri, Jun 19, 2009 at 3:27 PM, Bradley Brashier wrote: > So it turns out that it wasn't a bug at all -- it is a feature that was not > implemented. So I've got some work to do to get that running. Since I said I > would, though, here's my analysis of the conference parameters you were > asking about: > > mute-detect-sound > Different sound for if muting using only when mute-detect flag is on. > max-members > Specifies the maximum number of participants in a call. > max-members-sound > If caller cannot join because max is reached, this sound plays. Recommended > if max-members is set. > comfort-noise-level > Sets volume of background white noise to generate. > announce-count > Requires TTS. When joining, tells caller how many callers are already in > conference if at least the specified minimum. > suppress-events > ? Sets a flag, but does not appear to do anything with it. > I think suppress-events is to supress audio events such as "User XXXX has joined the conferece." Would the sound right? > > verbose-events > Maximum verbosity for transcripting. > timer-name > Specifies the name of this profile's timer. To separate it from other > timers? > > BB > On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier wrote: > >> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I >> chose flite), uncommenting a few things, and setting the TTS variables in >> the conference profile. You do have to rebuild FS to do this. >> >> With that working, my count function works, too. >> >> I posted a bug last night about conferencing, BTW: if you're using the >> wait-mod function, where the conference doesn't start until the moderator >> arrives, and you're also using separate profiles for users and moderators, >> the users only have the "user" profile until the first moderator arrives. At >> that time, they switch to also be using the "moderator" profile. >> >> Now that TTS is working, I'm going to see about helping you fill out those >> ???s, and maybe see if I can figure out how to fix the above bug. >> >> BB >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/cdc7d183/attachment-0002.html From msc at freeswitch.org Fri Jun 19 13:14:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:14:59 -0500 Subject: [Freeswitch-users] Quick heads up re: minor change in default configs Message-ID: <87f2f3b90906191314j2e301680sd4e0c16e1037cf34@mail.gmail.com> FYI, Just in case this affects you I wanted to give you all a heads up. With the new NAT busting code we found some stuff in the default configs that was no longer applicable and made some changes. Check them out: http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=13874 If you have any dialplan elements that make use of the "use_profile" channel variable please note that we moved it to vars.xml and set it to a default value of 'internal' since that works for just about everything. You can, of course, do whatever you want with it in your customizations. Happy FreeSWITCHing! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/e8fa125e/attachment-0002.html From msc at freeswitch.org Fri Jun 19 13:17:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:17:18 -0500 Subject: [Freeswitch-users] CHANNEL_HANGUP event no longer having variable_billsec in header In-Reply-To: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> References: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> Message-ID: <87f2f3b90906191317n6d47e07du2922a08eb3855b16@mail.gmail.com> Check out this change: http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=13505 Possibly you need to wait for the CHANNEL_HANGUP_COMPLETE event? -MC On Fri, Jun 19, 2009 at 2:15 PM, Matthew Fong wrote: > I upgraded to 13857 today, but noticed that the channel_hangup event no > longer contain the variable_billsec header. > Is this correct, or am I crazy? Thanks. > > --matt > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/eaee4a7d/attachment-0002.html From juanma.v82 at gmail.com Fri Jun 19 13:18:28 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Fri, 19 Jun 2009 17:18:28 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I have another switch who is in charge of it, witch is from another technology), I only want to negotiate the codecs in the way that I want it. This only seams to work when bypass media or proxy media is set to false. Due I need to use it as a SBC(session border controller) or pseudo proxy (I already know that is not intend for it), I need to negotiate the codecs in the FS. In the current thread I've already explained what I'm trying to do. If you give me a tip I'm willing to make the documentation richer. Thanks Regards In my architecture the switch who is in charge of transcoding IS NOT a FS. On 19/06/2009, at 16:19, Brian West wrote: > No right now you can not legally transcode G729 in FreeSWITCH, PERIOD! > > /b > > On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > >> Yes, it can do transcoding. Transcoding isn't the problem to my >> architecture, my problem is the codec negotiation between FS and >> Endpoints. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Fri Jun 19 13:19:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 19 Jun 2009 21:19:33 +0100 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: You are indeed correct, it's the 64bit server that performs well, not the 32bit PAE version. I'm hoping that's the cause. I need to dig around and find out if it's possible to change the kernel remotely and see it sorts the issue. Ultimately I'll update it to 64 bit anyway, but that's a 500 mile trek. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Burgess Sent: 19 June 2009 16:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] high cpu utilization On Fri, Jun 19, 2009 at 2:23 AM, Nik Middleton wrote: > I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I > don't think that's the issue I could be wrong, but I think PAE is a 32-bit kernel adapted for hardware with >4GB RAM. This can create a lot of overhead compared to running a true 64-bit kernel or a 32-bit kernel without PAE. Confirm this with 'uname -a' db _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From matt at hellohunter.com Fri Jun 19 13:21:06 2009 From: matt at hellohunter.com (Matt Hunter) Date: Fri, 19 Jun 2009 13:21:06 -0700 Subject: [Freeswitch-users] CHANNEL_HANGUP event no longer having variable_billsec in header In-Reply-To: <87f2f3b90906191317n6d47e07du2922a08eb3855b16@mail.gmail.com> References: <4256bf830906191215s762cdf81ha49fa2b0ba959e12@mail.gmail.com> <87f2f3b90906191317n6d47e07du2922a08eb3855b16@mail.gmail.com> Message-ID: <4256bf830906191321uf098ff5t62ee353b7c900611@mail.gmail.com> Thanks! On Fri, Jun 19, 2009 at 1:17 PM, Michael Collins wrote: > Check out this change: > http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=13505 > > Possibly you need to wait for the CHANNEL_HANGUP_COMPLETE event? > -MC > > On Fri, Jun 19, 2009 at 2:15 PM, Matthew Fong wrote: > >> I upgraded to 13857 today, but noticed that the channel_hangup event no >> longer contain the variable_billsec header. >> Is this correct, or am I crazy? Thanks. >> >> --matt >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/b6fac9e5/attachment-0002.html From bjbrashier at gmail.com Fri Jun 19 13:20:56 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Fri, 19 Jun 2009 13:20:56 -0700 Subject: [Freeswitch-users] Controlling Conference Controls In-Reply-To: <5a8712120906191305x5a98228eyb1847cfd2905aab5@mail.gmail.com> References: <7bcfdd290906171445i11530536na7aa5ad9f408738b@mail.gmail.com> <7bcfdd290906181715r41bd28d8rf828d6fd5bf57468@mail.gmail.com> <24104639.post@talk.nabble.com> <7bcfdd290906182030t7463dc71xd86e61e52c07a659@mail.gmail.com> <4A3B17F5.9070400@gmail.com> <7bcfdd290906182214odb793beu6bb85a5491e68556@mail.gmail.com> <20090619064325.GA22568@jdc.jasonjgw.net> <7bcfdd290906190859l3e70754cy807943494e693c53@mail.gmail.com> <7bcfdd290906191127x7654b580q482da30e512dd795@mail.gmail.com> <5a8712120906191305x5a98228eyb1847cfd2905aab5@mail.gmail.com> Message-ID: <7bcfdd290906191320m581dfe57sdb06404bd7e27ce3@mail.gmail.com> That sounds like something it might do, but searching through the code I didn't find any use of the flag that parameter sets. It is, of course, possible that I missed it, especially if it's an indirect use. 2009/6/19 Jo?o Mesquita > Inline... > > On Fri, Jun 19, 2009 at 3:27 PM, Bradley Brashier wrote: > >> So it turns out that it wasn't a bug at all -- it is a feature that was >> not implemented. So I've got some work to do to get that running. Since I >> said I would, though, here's my analysis of the conference parameters you >> were asking about: >> >> mute-detect-sound >> Different sound for if muting using only when mute-detect flag is on. >> max-members >> Specifies the maximum number of participants in a call. >> max-members-sound >> If caller cannot join because max is reached, this sound plays. >> Recommended if max-members is set. >> comfort-noise-level >> Sets volume of background white noise to generate. >> announce-count >> Requires TTS. When joining, tells caller how many callers are already in >> conference if at least the specified minimum. >> suppress-events >> ? Sets a flag, but does not appear to do anything with it. >> > > I think suppress-events is to supress audio events such as "User XXXX has > joined the conferece." > Would the sound right? > > >> >> verbose-events >> Maximum verbosity for transcripting. >> timer-name >> Specifies the name of this profile's timer. To separate it from other >> timers? >> >> BB >> On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier > > wrote: >> >>> OK, I figured out the TTS stuff. It's a matter of choosing an engine (I >>> chose flite), uncommenting a few things, and setting the TTS variables in >>> the conference profile. You do have to rebuild FS to do this. >>> >>> With that working, my count function works, too. >>> >>> I posted a bug last night about conferencing, BTW: if you're using the >>> wait-mod function, where the conference doesn't start until the moderator >>> arrives, and you're also using separate profiles for users and moderators, >>> the users only have the "user" profile until the first moderator arrives. At >>> that time, they switch to also be using the "moderator" profile. >>> >>> Now that TTS is working, I'm going to see about helping you fill out >>> those ???s, and maybe see if I can figure out how to fix the above bug. >>> >>> BB >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/b29dc545/attachment-0002.html From mattdfong at gmail.com Fri Jun 19 13:20:17 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 13:20:17 -0700 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory Message-ID: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> With yesterday's trunk and also a release from 2 weeks ago, I noticed that my freeswitch process as it ran was eating up more and more memory. At the end of the day it was using 75% of the sever's memory (About 12 gigs). It starts out taking a small amount of memory, and then throughout the day it slowly takes more and more. Is this normal? I'm using several lua ivr scripts...and have about 600-900 channels. Whats the best way to go about tracking down the cause? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/d8fabe9a/attachment-0002.html From brian at freeswitch.org Fri Jun 19 13:28:02 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 19 Jun 2009 15:28:02 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> Message-ID: Depends on what you're doing ... or not doing... /b On Jun 19, 2009, at 3:20 PM, Matthew Fong wrote: > With yesterday's trunk and also a release from 2 weeks ago, I > noticed that my freeswitch process as it ran was eating up more and > more memory. At the end of the day it was using 75% of the sever's > memory (About 12 gigs). It starts out taking a small amount of > memory, and then throughout the day it slowly takes more and more. > Is this normal? I'm using several lua ivr scripts...and have about > 600-900 channels. Whats the best way to go about tracking down the > cause? Thanks. > > --matt From anthony.minessale at gmail.com Fri Jun 19 13:38:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Jun 2009 15:38:43 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> Message-ID: <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> are you connecting to a db with the lua? On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: > With yesterday's trunk and also a release from 2 weeks ago, I noticed that > my freeswitch process as it ran was eating up more and more memory. At the > end of the day it was using 75% of the sever's memory (About 12 gigs). It > starts out taking a small amount of memory, and then throughout the day it > slowly takes more and more. Is this normal? I'm using several lua ivr > scripts...and have about 600-900 channels. Whats the best way to go about > tracking down the cause? Thanks. > --matt > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/ab0ccfdc/attachment-0002.html From msc at freeswitch.org Fri Jun 19 13:40:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:40:17 -0500 Subject: [Freeswitch-users] CTI In-Reply-To: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> Message-ID: <87f2f3b90906191340l14e2e0adx2360497b6ee5b464@mail.gmail.com> I'm not aware of any pre-existing software like this. FreeSWITCH has all the hooks for someone to create the abstraction layers for CSTA, TAPI, VoiceXML, etc. but no one has ponied up the money to pay for the development... -MC 2009/6/18 Maxim Tsvetov > Hello! > > We are seeking possibilities to use CTI features with Freeswitch. > > This features are: > - click-to-dial > - call popup > - answer call,hangup > - call transfer > > > Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..) > or there is already written module or third-party software? > This solution should support 100-150 simultaneous ?onnections from > freeswitch users. > > Could you please share you experience with CTI. > > Regards, > Maxim Tsvetov > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/1e2df734/attachment-0002.html From msc at freeswitch.org Fri Jun 19 13:44:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:44:50 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> Message-ID: <87f2f3b90906191344y65a7943bs8c953007f4c95f91@mail.gmail.com> On another email thread Tony mentioned using "top -H" and then doing a gcore to locate the process(es) that are sucking up all the resources. Have you been down that path already? -MC On Fri, Jun 19, 2009 at 3:28 PM, Brian West wrote: > Depends on what you're doing ... or not doing... > > /b > > On Jun 19, 2009, at 3:20 PM, Matthew Fong wrote: > > > With yesterday's trunk and also a release from 2 weeks ago, I > > noticed that my freeswitch process as it ran was eating up more and > > more memory. At the end of the day it was using 75% of the sever's > > memory (About 12 gigs). It starts out taking a small amount of > > memory, and then throughout the day it slowly takes more and more. > > Is this normal? I'm using several lua ivr scripts...and have about > > 600-900 channels. Whats the best way to go about tracking down the > > cause? Thanks. > > > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/6508e66c/attachment-0002.html From mattdfong at gmail.com Fri Jun 19 13:53:57 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 19 Jun 2009 13:53:57 -0700 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> Message-ID: <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> some lua event listeners are connecting to mysql with lua..but the connection is created once, and kept open the lua ivr's do *not *connect to any database. top -H seems to show an even distribution of of cpu and memory usage amongst freeswitch threads. Nothing seems out of the ordinary with a specific thread. --matt On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > are you connecting to a db with the lua? > > > On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: > >> With yesterday's trunk and also a release from 2 weeks ago, I noticed that >> my freeswitch process as it ran was eating up more and more memory. At the >> end of the day it was using 75% of the sever's memory (About 12 gigs). It >> starts out taking a small amount of memory, and then throughout the day it >> slowly takes more and more. Is this normal? I'm using several lua ivr >> scripts...and have about 600-900 channels. Whats the best way to go about >> tracking down the cause? Thanks. >> --matt >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/7f5a0c59/attachment-0002.html From msc at freeswitch.org Fri Jun 19 13:55:04 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:55:04 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> Message-ID: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Now I feel stupid because I didn't read your original post closely enough. You've defined your "mysocket" extension in the "public" context but when you do an origination with sofia/internal/foo at bar it will use the "default" context. I think the quickest way to handle this is to create a copy of your mysocket.xml file and put it in conf/dialplan/default/ and be done with it. -MC On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater wrote: > Mike, > > Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to > me though. > > Max. > > > On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: > >> Max, >> that pastebin failed miserably as none of the xml shows up. can you try >> again or use our pastebin.freeswitch.org site? >> -MC >> >> >> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Hi Mike, >>> >>> It's pasted here: http://pastebin.ca/1466521 >>> >>> Thanks, >>> Max. >>> >>> >>> >>> >>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >>> >>>> Can you turn on debugging (F8) and capture all the output after your >>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>> -MC >>>> >>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>> max.bridgewater at gmail.com> wrote: >>>> >>>>> Any help our there? >>>>> >>>>> I'm still trying to get this piece working. Essentially what i wan to >>>>> do is, when a call comes in (from registered devices as well as unregistered >>>>> devices), notify the my server socket. Somehow it's not working. The change >>>>> i made compared to the standard Freeswitch settings are the following: >>>>> >>>>> 1) Added following extension that in >>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2) Changed file: >>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I noticed that with this extension, all calls received from external >>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>> But calls from registered devices and initiated using the socket interface >>>>> are not forwarded. Is there something that need to be changed in the >>>>> profiles? >>>>> >>>>> or is something wrong with my dial string? >>>>> {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62. >>>>> >>>>> >>>>> In the logs, i cannot see that that my extension is being matched. >>>>> >>>>> Any idea, >>>>> >>>>> Max. >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/84e3ac62/attachment-0002.html From msc at freeswitch.org Fri Jun 19 13:59:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Jun 2009 15:59:02 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> Message-ID: <87f2f3b90906191359t179fc74bw74e24abca3c4f42c@mail.gmail.com> Can you pastebin your script and your dialplan? There might be some clue there.. On Fri, Jun 19, 2009 at 3:53 PM, Matthew Fong wrote: > some lua event listeners are connecting to mysql with lua..but the > connection is created once, and kept open > the lua ivr's do *not *connect to any database. > > top -H seems to show an even distribution of of cpu and memory usage > amongst freeswitch threads. Nothing seems out of the ordinary with a > specific thread. > > --matt > On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> are you connecting to a db with the lua? >> >> >> On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: >> >>> With yesterday's trunk and also a release from 2 weeks ago, I noticed >>> that my freeswitch process as it ran was eating up more and more memory. At >>> the end of the day it was using 75% of the sever's memory (About 12 gigs). >>> It starts out taking a small amount of memory, and then throughout the day >>> it slowly takes more and more. Is this normal? I'm using several lua ivr >>> scripts...and have about 600-900 channels. Whats the best way to go about >>> tracking down the cause? Thanks. >>> --matt >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/762d3a78/attachment-0002.html From anthony.minessale at gmail.com Fri Jun 19 14:07:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Jun 2009 16:07:37 -0500 Subject: [Freeswitch-users] Diagnosing FreeSWITCH eating more and more memory In-Reply-To: <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> References: <4256bf830906191320o4e9f4475m20006f88b812c3a6@mail.gmail.com> <191c3a030906191338q4427dc09h7f851266eb58df1a@mail.gmail.com> <4256bf830906191353h1025bb11o7541d6a6017281f1@mail.gmail.com> Message-ID: <191c3a030906191407v1eab84bes7f61704a2ef77e4@mail.gmail.com> try removing certian elements of you setup to narrrow it down one at a time. remove the lua + sql, the ivr scripts etc and see if you can pinpoint your problem it's amost for sure going to be in lua and probably some plugin for it. On Fri, Jun 19, 2009 at 3:53 PM, Matthew Fong wrote: > some lua event listeners are connecting to mysql with lua..but the > connection is created once, and kept open > the lua ivr's do *not *connect to any database. > > top -H seems to show an even distribution of of cpu and memory usage > amongst freeswitch threads. Nothing seems out of the ordinary with a > specific thread. > > --matt > On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> are you connecting to a db with the lua? >> >> >> On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote: >> >>> With yesterday's trunk and also a release from 2 weeks ago, I noticed >>> that my freeswitch process as it ran was eating up more and more memory. At >>> the end of the day it was using 75% of the sever's memory (About 12 gigs). >>> It starts out taking a small amount of memory, and then throughout the day >>> it slowly takes more and more. Is this normal? I'm using several lua ivr >>> scripts...and have about 600-900 channels. Whats the best way to go about >>> tracking down the cause? Thanks. >>> --matt >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/29fb08d4/attachment-0002.html From apt.get at gmail.com Fri Jun 19 14:09:13 2009 From: apt.get at gmail.com (David Burgess) Date: Fri, 19 Jun 2009 15:09:13 -0600 Subject: [Freeswitch-users] high cpu utilization In-Reply-To: References: <4cd9d780906181330s35e1dbc6g1884389d0d000c1f@mail.gmail.com> Message-ID: On Fri, Jun 19, 2009 at 2:19 PM, Nik Middleton wrote: > You are indeed correct, it's the 64bit server that performs well, not > the 32bit PAE version. ?I'm hoping that's the cause. ?I need to dig > around and find out if it's possible to change the kernel remotely and > see it sorts the issue. ?Ultimately I'll update it to 64 bit anyway, but > that's a 500 mile trek. I don't think it's as simple as changing the kernel. My understanding is that when you change arch you are reinstalling the system. Given that 32-bit binaries and libs generally will run in a 64-bit environment though, there may be a way to swing it. I'm not sure how you would determine whether that were really your problem. No question a PAE kernel does create overhead though, in some situations more than others. db From max.bridgewater at gmail.com Fri Jun 19 14:22:33 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Fri, 19 Jun 2009 17:22:33 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Message-ID: I don't have my settings to try it right now. Still i have a question. If it's the way you describe it, why wouldn't sofia/extenal/foo at bar solve the problem? I think i even copied the extension both to the default directory. But i will confirm and let you know. Max. On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins wrote: > Now I feel stupid because I didn't read your original post closely enough. > > > You've defined your "mysocket" extension in the "public" context but when > you do an origination with sofia/internal/foo at bar it will use the > "default" context. I think the quickest way to handle this is to create a > copy of your mysocket.xml file and put it in conf/dialplan/default/ and be > done with it. > > -MC > > > On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Mike, >> >> Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to >> me though. >> >> Max. >> >> >> On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: >> >>> Max, >>> that pastebin failed miserably as none of the xml shows up. can you try >>> again or use our pastebin.freeswitch.org site? >>> -MC >>> >>> >>> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >>> max.bridgewater at gmail.com> wrote: >>> >>>> Hi Mike, >>>> >>>> It's pasted here: http://pastebin.ca/1466521 >>>> >>>> Thanks, >>>> Max. >>>> >>>> >>>> >>>> >>>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >>>> >>>>> Can you turn on debugging (F8) and capture all the output after your >>>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>>> -MC >>>>> >>>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>>> max.bridgewater at gmail.com> wrote: >>>>> >>>>>> Any help our there? >>>>>> >>>>>> I'm still trying to get this piece working. Essentially what i wan to >>>>>> do is, when a call comes in (from registered devices as well as unregistered >>>>>> devices), notify the my server socket. Somehow it's not working. The change >>>>>> i made compared to the standard Freeswitch settings are the following: >>>>>> >>>>>> 1) Added following extension that in >>>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2) Changed file: >>>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> I noticed that with this extension, all calls received from external >>>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>>> But calls from registered devices and initiated using the socket interface >>>>>> are not forwarded. Is there something that need to be changed in the >>>>>> profiles? >>>>>> >>>>>> or is something wrong with my dial string? >>>>>> {origination_caller_id_number=12000}sofia/internal/ >>>>>> 242424 at 192.168.1.62. >>>>>> >>>>>> In the logs, i cannot see that that my extension is being matched. >>>>>> >>>>>> Any idea, >>>>>> >>>>>> Max. >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/5b80a16d/attachment-0002.html From timb0311 at hotmail.com Fri Jun 19 17:52:31 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 20:52:31 -0400 Subject: [Freeswitch-users] Update - Transmit fax locally for test In-Reply-To: References: Message-ID: Yes that is the extension defined in the default dialplan. It is setup like explained under mod_fax... here is the actual definition: http://pastebin.freeswitch.org/9450 > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503 at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000 at x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users at lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > _________________________________________________________________ Hotmail? has ever-growing storage! Don?t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutorial_Storage_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/9ab0f893/attachment-0002.html From timb0311 at hotmail.com Fri Jun 19 20:06:10 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 19 Jun 2009 23:06:10 -0400 Subject: [Freeswitch-users] (Found Fix) Transmit fax locally for test In-Reply-To: References: Message-ID: Ok so after many attempts of trial and error I narrowed it down to acls. So when trying to orginate a call to the local FS extension it was getting blocked. Adding the following allow with my freeswitch IP to the domains list allowed the originate to take place. acl.conf.xml: So now this statement works for local fax testing: originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) Now my question is, is this the proper or best way to configure this? Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503 at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000 at x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users at lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > > haha, yeah i see it now... duh. pulled an all nighter, too many things > going on. must have overlooked it. > > > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > > the login and password is pastebin/freeswitch > > > > been trying to break myself into freeswitch on top of my original workload. > thanks for the help. > _________________________________________________________________ Bing? brings you maps, menus, and reviews organized in one place. Try it now. http://www.bing.com/search?q=restaurants&form=MLOGEN&publ=WLHMTAG&crea=TEXT_MLOGEN_Core_tagline_local_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090619/f05205dd/attachment-0002.html From mariano.dellano at gmail.com Fri Jun 19 14:28:04 2009 From: mariano.dellano at gmail.com (Mariano de Llano) Date: Fri, 19 Jun 2009 18:28:04 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: You are right, it seams that it can not be done. In the past I've tried to do something similiar but with no success. Apperently the documentation is wrong. If I have some time I will look at the code and I will give you some feedback. Cheers, PS: The transcoding question has nothing to do with your question :) On 19/06/2009, at 17:18, JuanMa wrote: > I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I > have another switch who is in charge of it, witch is from another > technology), I only want to negotiate the codecs in the way that I > want it. This only seams to work when bypass media or proxy media is > set to false. Due I need to use it as a SBC(session border controller) > or pseudo proxy (I already know that is not intend for it), I need to > negotiate the codecs in the FS. In the current thread I've already > explained what I'm trying to do. If you give me a tip I'm willing to > make the documentation richer. > > Thanks > Regards > > In my architecture the switch who is in charge of transcoding IS NOT a > FS. > > > On 19/06/2009, at 16:19, Brian West wrote: > >> No right now you can not legally transcode G729 in FreeSWITCH, >> PERIOD! >> >> /b >> >> On Jun 19, 2009, at 2:11 PM, JuanMa wrote: >> >>> Yes, it can do transcoding. Transcoding isn't the problem to my >>> architecture, my problem is the codec negotiation between FS and >>> Endpoints. >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Jun 19 21:44:30 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 20 Jun 2009 00:44:30 -0400 Subject: [Freeswitch-users] Mod_Fax / TxFax / Originate In-Reply-To: References: Message-ID: <6D803586-B8E1-4EAB-A780-6A3EAD680C5F@jerris.com> Try using loopback endpoint for this test . Mike On Jun 17, 2009, at 10:00 AM, Tim B wrote: > Trying to do a local test for faxing. Keep getting an error. > > default dialplan: > > > > > > > > > > > //inbound from remote box works fine > - connect asterisk box and fs box, then fax from asterisk to fs... OK > - also fax from fs to asterisk.... OK > > // local fax on fs .... FAILS!! > // my originate command: > originate sofia/internal/8000 at 192.168.10.35 &txfax(storage/fax/ > test.tif) > > // error as follows: > 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing > FreeSWITCH->8000 in context public > 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer > 1 Legged calls > 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000 at 192.168.10.35 > [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > > > > > Lauren found her dream laptop. Find the PC that?s right for you. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/d21742bc/attachment-0002.html From mike at jerris.com Fri Jun 19 21:46:59 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 20 Jun 2009 00:46:59 -0400 Subject: [Freeswitch-users] Freeswitch as a B2B Application Server for IMS In-Reply-To: References: Message-ID: Some of these things make sense is some scenarios but not others. Most people are wanting to do full topology hiding, so we don't by default pass very much across a bridge. I am interested in working on this, feel free to contact me off list with your findings. Mike On Jun 17, 2009, at 11:20 AM, Cavalera Claudio Luigi wrote: > Hello freeswitchers, > lately I'm trying to set up a testbed to ivestigate a potential use of > freeswitch as a Back-to-Back application server in an IMS > architecture. > > I've seen IMS specs are also linked here > http://wiki.freeswitch.org/wiki/Documentation > so I've thinked maybe there's a chance :-) > > I also have the inner feeling that fs could do an amazing job in IMS > field as he does in NGN. > > For my testing I'm now using OpenIMSCore as control layer (where > phones > do register) and I'm trying to put fs on top of it as a b2b > application > server to provide services. > > I would like to share with you my experience to see if something could > be done about this scenario (or if it's worth the trouble at least). > > Alice and Bob are two users registered to the IMS Core and they both > have a profile for which originating and terminating INVITEs get > triggered towards the fs application server. > > When Alice calls Bob the call setup would include three legs: > > 1) Alice -> PCSCF (orig) -> SCSCF (orig) -> FS (orig) > > 2) FS (orig) -> SCSCF (orig) -> ICSCF -> SCSCF (term) -> FS (term) > > 3) FS (term) -> SCSCF (term) -> PCSCF(term) -> Bob > > > This partially works already with fs out of the box, but there are a > still a few issues to be solved. > > When FS starts the brand new leg 2) as a B2B User Agent he should keep > the Route: header in the SIP INVITE "almost the same" as the one he > received in the leg 1) > > I see here two different issues: > a) Getting the Route: header out of incoming invite in leg 1) > b) Writing the proper Route: header and have FS behaving correctly at > transport layer in the outgoing INVITE in leg 2) > > a) Now please correct me if I'm wrong: at the moment the > header > is not a channel variable available in fs (e.g. I don't get it with > the > "info" app). It there were a way to get this header out of the > incoming > INVITEs, I could do the logic to parse it and forge a proper one in > the > outgoing INVITE. > > > b) Concerning how to write the header, I'm already working > with > fs_path directive which also makes FS behaves correctly at network > layer. > Could someone please elaborate a little bit about the alternative to > fs_path directive? > > I've seen there are already many in theory: > - combining sip_h_Route= with > http://wiki.freeswitch.org/wiki/Variable_sip_network_destination > - use of http://wiki.freeswitch.org/wiki/Variable_sip_route_uri > - use of fs_path= > http://wiki.freeswitch.org/wiki/ > Sofia#Specifying_SIP_Proxy_With_fs_path > > > I've simplified the scenario a little bit, there are other things that > the B2B AS should do (e.g. removing Record-Route:) but FS do them > already from what I've tested. > If anyone in the community is interested I'm here to provide further > information or share my experience to implement the best solution. > > Best regards, > Claudio > > > > > Internet Email Confidentiality Footer > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e > ogni documento o file allegato, e' rivolta unicamente alla/e persona/ > e cui e' indirizzata ed alle altre da questa autorizzata/e a > riceverla. Se non siete i destinatari/autorizzati siete avvisati che > qualsiasi azione, copia, comunicazione, divulgazione o simili basate > sul contenuto di tali informazioni e' vietata e potrebbe essere > contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia > di protezione dei dati personali). Se avete ricevuto questa > comunicazione per errore, vi preghiamo di darne immediata notizia al > mittente e di distruggere il messaggio originale e ogni file > allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) > only and are confidential and/or may contain legally privileged > information. If you have received this message by mistake or are not > one of the addressees above, you may take no action based on it, and > you may not copy or show it to anyone; please reply to this e-mail > and point out the error which has occurred. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bipin at xbipin.com Fri Jun 19 22:42:08 2009 From: bipin at xbipin.com (xbipin) Date: Fri, 19 Jun 2009 22:42:08 -0700 (PDT) Subject: [Freeswitch-users] need to get some basics right Message-ID: <24122294.post@talk.nabble.com> im just confused about certains tuff in freeswitch, firstly if we use TLS for SIP then FS can read such packets but if it were to act as gateway can it communicate in plain SIP, so its like phone connected to FS using TLS and then from FS to gateway in plain SIP? secondly, if phone uses SRTP, can FS take that media stream and convert to plain RTP so it can forward it to remote Gateway, i know it cant do for G723, G729 due to license issues but can it do for the other codecs? my main concern is can it actually act as a encryption and decryption server by remaining between the gateway and phone? -- View this message in context: http://www.nabble.com/need-to-get-some-basics-right-tp24122294p24122294.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mrene_lists at avgs.ca Fri Jun 19 22:44:28 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 20 Jun 2009 00:44:28 -0500 Subject: [Freeswitch-users] need to get some basics right In-Reply-To: <24122294.post@talk.nabble.com> References: <24122294.post@talk.nabble.com> Message-ID: You can do all that. SRTP<>RTP doesnt depend on the codec since it just decrypts the packet's contents and SIP/TLS is dependent on that specific connection. Math On 20-Jun-09, at 12:42 AM, xbipin wrote: > > im just confused about certains tuff in freeswitch, firstly if we > use TLS for > SIP then FS can read such packets but if it were to act as gateway > can it > communicate in plain SIP, so its like phone connected to FS using > TLS and > then from FS to gateway in plain SIP? > secondly, if phone uses SRTP, can FS take that media stream and > convert to > plain RTP so it can forward it to remote Gateway, i know it cant do > for > G723, G729 due to license issues but can it do for the other codecs? > > my main concern is can it actually act as a encryption and > decryption server > by remaining between the gateway and phone? > -- > View this message in context: http://www.nabble.com/need-to-get-some-basics-right-tp24122294p24122294.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raffaele.p.guidi at gmail.com Sat Jun 20 05:40:56 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sat, 20 Jun 2009 14:40:56 +0200 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3AF28D.4040708@gmail.com> References: <4A3AF28D.4040708@gmail.com> Message-ID: uhm... nice! But why not MPL license (the same as freeswitch)? On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com wrote: > http://versafon.com/versafonweb/Software.jsp > > Essentially it's a wrapper around inbound socket interface, not all > events supported yet, and not all event parameters/variables. It's multi > threaded and scaled well in testing. > We offer commercial support and development for FreeSwitch as well. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/defe058c/attachment-0002.html From diego.viola at gmail.com Sat Jun 20 06:00:37 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 20 Jun 2009 09:00:37 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: References: <4A3AF28D.4040708@gmail.com> Message-ID: <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> Because they probably want a stronger copyleft license? I prefer the GPL because of that reason. On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > uhm... nice! But why not MPL license (the same as freeswitch)? > > > On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com wrote: > >> http://versafon.com/versafonweb/Software.jsp >> >> Essentially it's a wrapper around inbound socket interface, not all >> events supported yet, and not all event parameters/variables. It's multi >> threaded and scaled well in testing. >> We offer commercial support and development for FreeSwitch as well. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/7c39acb0/attachment-0002.html From paul.degt at gmail.com Sat Jun 20 07:37:59 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 10:37:59 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> Message-ID: <4A3CF447.3030501@gmail.com> Yes, that's one of the reasons. Another point is that GPL v.3 is defined more clearly from legal perspective, at least from our legal adviser point of view. Diego Viola wrote: > Because they probably want a stronger copyleft license? > > I prefer the GPL because of that reason. > > On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > > wrote: > > uhm... nice! But why not MPL license (the same as freeswitch)? > > > On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > > wrote: > > http://versafon.com/versafonweb/Software.jsp > > Essentially it's a wrapper around inbound socket interface, > not all > events supported yet, and not all event parameters/variables. > It's multi > threaded and scaled well in testing. > We offer commercial support and development for FreeSwitch as > well. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From saeedahmad1981 at gmail.com Sat Jun 20 08:06:35 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Sat, 20 Jun 2009 17:06:35 +0200 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: Hi, I am not expert, and if i understood you correctly, i am doing it like that: set late negotiation to false (it can be set true if you want to negotiate it on the fly, but it didn't work for me) don't use absolute_codec_string instead use codec_string and put codec(s) what EP2 is offering, now if A side is sending one of the codec which are defined for EP2 then it will be used. i didn't consider your FS at right side because your SWITCH definately going to transcode the codec into G729, otherwise above scenario should be same for both FS. let me know if it also works for you, i am also using it in proxy media mode. are you using xml_curl? - Saeed On Fri, Jun 19, 2009 at 11:28 PM, Mariano de Llano < mariano.dellano at gmail.com> wrote: > > You are right, it seams that it can not be done. > > In the past I've tried to do something similiar but with no success. > Apperently the documentation is wrong. > > If I have some time I will look at the code and I will give you some > feedback. > > Cheers, > > PS: The transcoding question has nothing to do with your question :) > > On 19/06/2009, at 17:18, JuanMa wrote: > > > I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I > > have another switch who is in charge of it, witch is from another > > technology), I only want to negotiate the codecs in the way that I > > want it. This only seams to work when bypass media or proxy media is > > set to false. Due I need to use it as a SBC(session border controller) > > or pseudo proxy (I already know that is not intend for it), I need to > > negotiate the codecs in the FS. In the current thread I've already > > explained what I'm trying to do. If you give me a tip I'm willing to > > make the documentation richer. > > > > Thanks > > Regards > > > > In my architecture the switch who is in charge of transcoding IS NOT a > > FS. > > > > > > On 19/06/2009, at 16:19, Brian West wrote: > > > >> No right now you can not legally transcode G729 in FreeSWITCH, > >> PERIOD! > >> > >> /b > >> > >> On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > >> > >>> Yes, it can do transcoding. Transcoding isn't the problem to my > >>> architecture, my problem is the codec negotiation between FS and > >>> Endpoints. > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/98688012/attachment-0002.html From brian at freeswitch.org Sat Jun 20 08:26:27 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 20 Jun 2009 10:26:27 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3CF447.3030501@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> Message-ID: <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> I still say why not MPL or at the very least MPL/GPL? /b On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > Yes, that's one of the reasons. Another point is that GPL v.3 is > defined > more clearly from legal perspective, at least from our legal adviser > point of view. > > Diego Viola wrote: >> Because they probably want a stronger copyleft license? >> >> I prefer the GPL because of that reason. >> >> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi >> > >> wrote: >> >> uhm... nice! But why not MPL license (the same as freeswitch)? >> >> >> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com >> > > wrote: >> >> http://versafon.com/versafonweb/Software.jsp >> >> Essentially it's a wrapper around inbound socket interface, >> not all >> events supported yet, and not all event parameters/variables. >> It's multi >> threaded and scaled well in testing. >> We offer commercial support and development for FreeSwitch as >> well. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul.degt at gmail.com Sat Jun 20 09:12:15 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 12:12:15 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> Message-ID: <4A3D0A5F.4080608@gmail.com> As I mentioned the decision was influenced by our legal adviser. And it probably can be relicensed as MPL/GPL, but why? My understanding is that for an end user which wants to use a free open source software there should be absolutely no difference between GPL and MPL. But if any one has has any licensing concern with his particular use case - please contact me with the details, I will do what I can to help. Brian West wrote: > I still say why not MPL or at the very least MPL/GPL? > > /b > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is >> defined >> more clearly from legal perspective, at least from our legal adviser >> point of view. >> >> Diego Viola wrote: >> >>> Because they probably want a stronger copyleft license? >>> >>> I prefer the GPL because of that reason. >>> >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi >>> > >>> wrote: >>> >>> uhm... nice! But why not MPL license (the same as freeswitch)? >>> >>> >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com >>> >> > wrote: >>> >>> http://versafon.com/versafonweb/Software.jsp >>> >>> Essentially it's a wrapper around inbound socket interface, >>> not all >>> events supported yet, and not all event parameters/variables. >>> It's multi >>> threaded and scaled well in testing. >>> We offer commercial support and development for FreeSwitch as >>> well. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sat Jun 20 09:29:21 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 20 Jun 2009 11:29:21 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D0A5F.4080608@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> Message-ID: <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> There actually are issues between the GPL and MPL :P /b On Jun 20, 2009, at 11:12 AM, paul.degt wrote: > source software there should be absolutely no difference between GPL > and > MPL. From paul.degt at gmail.com Sat Jun 20 10:07:18 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 13:07:18 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> Message-ID: <4A3D1746.9050504@gmail.com> Not unless you combining GPL and MPL code in one binary, which I find highly improbable in case of FreeSwitch. And even in this case there seems to be a workaround: "However, MPL 1.1 has a provision (section 13) that allows a program (or parts of it) to offer a choice of another license as well. If part of a program allows the GNU GPL as an alternate choice, or any other GPL-compatible license as an alternate choice, that part of the program has a GPL-compatible license." http://www.fsf.org/licensing/licenses/index_html#GPLIncompatibleLicenses Brian West wrote: > There actually are issues between the GPL and MPL :P > > /b > > On Jun 20, 2009, at 11:12 AM, paul.degt wrote: > > >> source software there should be absolutely no difference between GPL >> and >> MPL. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Sat Jun 20 10:21:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 20 Jun 2009 12:21:15 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D1746.9050504@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <97BAED85-F387-4263-B442-2C67DA920649@freeswitch.org> <4A3D1746.9050504@gmail.com> Message-ID: <191c3a030906201021o74aa892at68bef899089bd847@mail.gmail.com> License is less important in a socket interface lib. Its not really a big deal what license it is guys. However, I did create a BSD ESL lib distributed with FS to make sure nobody would have any license problems connecting to our stuff. If you don't like this lib consider using swig on the wrapper lib in libesl like I have done for perl, lua, php, python, ruby etc. On Jun 20, 2009 12:15 PM, "paul.degt" wrote: Not unless you combining GPL and MPL code in one binary, which I find highly improbable in case of FreeSwitch. And even in this case there seems to be a workaround: "However, MPL 1.1 has a provision (section 13) that allows a program (or parts of it) to offer a choice of another license as well. If part of a program allows the GNU GPL as an alternate choice, or any other GPL-compatible license as an alternate choice, that part of the program has a GPL-compatible license." http://www.fsf.org/licensing/licenses/index_html#GPLIncompatibleLicenses Brian West wrote: > There actually are issues between the GPL and MPL :P > > /b > > On Jun 20, 200... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/99398f40/attachment-0002.html From d at d-man.org Sat Jun 20 12:32:31 2009 From: d at d-man.org (Darren Schreiber) Date: Sat, 20 Jun 2009 12:32:31 -0700 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 Message-ID: <2C6A726363C34E2DAE22EF723C668E1C@test> Hello folks, An important stability issue was identified in the FreeSWITCH core ODBC drivers when utilizing decimal / float columns in databases (at least MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you are likely using floating point or decimal columns to track cash amounts. FreeSWITCH may periodically core dump when a floating point value is retrieved from the database due to this bug. Please upgrade to rev 13866 (or at least apply the patch - it can be applied independently if you are on at least rev 12152). You do not need to update mod_nibblebill, as the bug is not actually in mod_nibblebill, it just happens to cause the right conditions to exist to exhibit this behavior. Background information is in FSCORE-384 . Thanks to Tony for acting quickly on this patch. - Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/9dd200b0/attachment-0002.html From raffaele.p.guidi at gmail.com Sat Jun 20 12:39:34 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sat, 20 Jun 2009 21:39:34 +0200 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D0A5F.4080608@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> Message-ID: Well, GPL for a library is a dead end. Nobody will use it unless for open source projects. I would consider any possible choice before using it. Regards, Raffaele On Sat, Jun 20, 2009 at 18:12, paul.degt wrote: > As I mentioned the decision was influenced by our legal adviser. And it > probably can be relicensed as MPL/GPL, but why? > My understanding is that for an end user which wants to use a free open > source software there should be absolutely no difference between GPL and > MPL. > But if any one has has any licensing concern with his particular use > case - please contact me with the details, I will do what I can to help. > > > Brian West wrote: > > I still say why not MPL or at the very least MPL/GPL? > > > > /b > > > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > > > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is > >> defined > >> more clearly from legal perspective, at least from our legal adviser > >> point of view. > >> > >> Diego Viola wrote: > >> > >>> Because they probably want a stronger copyleft license? > >>> > >>> I prefer the GPL because of that reason. > >>> > >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > >>> > > >>> wrote: > >>> > >>> uhm... nice! But why not MPL license (the same as freeswitch)? > >>> > >>> > >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > >>> >>> > wrote: > >>> > >>> http://versafon.com/versafonweb/Software.jsp > >>> > >>> Essentially it's a wrapper around inbound socket interface, > >>> not all > >>> events supported yet, and not all event parameters/variables. > >>> It's multi > >>> threaded and scaled well in testing. > >>> We offer commercial support and development for FreeSwitch as > >>> well. > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > ------------------------------------------------------------------------ > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/321913f5/attachment-0002.html From diego.viola at gmail.com Sat Jun 20 12:41:11 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 20 Jun 2009 15:41:11 -0400 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 In-Reply-To: <2C6A726363C34E2DAE22EF723C668E1C@test> References: <2C6A726363C34E2DAE22EF723C668E1C@test> Message-ID: <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> Is this issue fixed in latest trunk? On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber wrote: > Hello folks, > An important stability issue was identified in the FreeSWITCH core > ODBC drivers when utilizing decimal / float columns in databases (at least > MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you > are likely using floating point or decimal columns to track cash amounts. > FreeSWITCH may periodically core dump when a floating point value is > retrieved from the database due to this bug. > > Please upgrade to rev 13866 (or at least apply the patch - > it can be applied independently if you are on at least rev 12152). You do > not need to update mod_nibblebill, as the bug is not actually in > mod_nibblebill, it just happens to cause the right conditions to exist to > exhibit this behavior. > > Background information is in FSCORE-384 > . > > Thanks to Tony for acting quickly on this patch. > > - Darren > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/3fdb9ffe/attachment-0002.html From d at d-man.org Sat Jun 20 12:44:26 2009 From: d at d-man.org (Darren Schreiber) Date: Sat, 20 Jun 2009 12:44:26 -0700 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 In-Reply-To: <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> References: <2C6A726363C34E2DAE22EF723C668E1C@test> <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> Message-ID: <113AC34E3E23432D8B7F87FAC3CC40A2@test> Yes _____ From: Diego Viola [mailto:diego.viola at gmail.com] Sent: Saturday, June 20, 2009 12:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] If you use mod_nibblebill,please upgrade to rev 13866 Is this issue fixed in latest trunk? On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber wrote: Hello folks, An important stability issue was identified in the FreeSWITCH core ODBC drivers when utilizing decimal / float columns in databases (at least MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you are likely using floating point or decimal columns to track cash amounts. FreeSWITCH may periodically core dump when a floating point value is retrieved from the database due to this bug. Please upgrade to rev 13866 (or at least apply the patch - it can be applied independently if you are on at least rev 12152). You do not need to update mod_nibblebill, as the bug is not actually in mod_nibblebill, it just happens to cause the right conditions to exist to exhibit this behavior. Background information is in FSCORE-384 . Thanks to Tony for acting quickly on this patch. - Darren _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/8283ad13/attachment-0002.html From diego.viola at gmail.com Sat Jun 20 12:45:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 20 Jun 2009 15:45:47 -0400 Subject: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866 In-Reply-To: <113AC34E3E23432D8B7F87FAC3CC40A2@test> References: <2C6A726363C34E2DAE22EF723C668E1C@test> <86a32abc0906201241q4094340fl33e6a35256a54641@mail.gmail.com> <113AC34E3E23432D8B7F87FAC3CC40A2@test> Message-ID: <86a32abc0906201245r43de8aboe17103541fab9682@mail.gmail.com> Cool, thanks, you guys are fast like the light ;) On Sat, Jun 20, 2009 at 3:44 PM, Darren Schreiber wrote: > Yes > > ------------------------------ > *From:* Diego Viola [mailto:diego.viola at gmail.com] > *Sent:* Saturday, June 20, 2009 12:41 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] If you use mod_nibblebill,please upgrade > to rev 13866 > > Is this issue fixed in latest trunk? > > On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber wrote: > >> Hello folks, >> An important stability issue was identified in the FreeSWITCH core >> ODBC drivers when utilizing decimal / float columns in databases (at least >> MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you >> are likely using floating point or decimal columns to track cash amounts. >> FreeSWITCH may periodically core dump when a floating point value is >> retrieved from the database due to this bug. >> >> Please upgrade to rev 13866 (or at least apply the patch - >> it can be applied independently if you are on at least rev 12152). You do >> not need to update mod_nibblebill, as the bug is not actually in >> mod_nibblebill, it just happens to cause the right conditions to exist to >> exhibit this behavior. >> >> Background information is in FSCORE-384 >> . >> >> Thanks to Tony for acting quickly on this patch. >> >> - Darren >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/58bc1057/attachment-0002.html From evilla at chipoly.com Sat Jun 20 12:51:48 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Sat, 20 Jun 2009 13:51:48 -0600 Subject: [Freeswitch-users] stucked with mod_fsv and mod_h26x Message-ID: <009001c9f1e0$8d1df990$a759ecb0$@com> Hello friends. i'm kind of stucked with mod_fsv and mod_h26x... there is no info on wiki yet. have u used them? Can u send examples of how-to? I need to join 2 endpoints with full HD video conference using PZT HD VideoCams to PC and SIP-UA. Thank you! EDWIN ChiPoLy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/14cf2161/attachment-0002.html From paul.degt at gmail.com Sat Jun 20 13:15:40 2009 From: paul.degt at gmail.com (paul.degt) Date: Sat, 20 Jun 2009 16:15:40 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> Message-ID: <4A3D436C.6030405@gmail.com> That's rather bold statement and indicates lack of knowledge on the subject . I would advise you to look at http://www.fsf.org/licensing/licenses/gpl-faq.html on GPL information. In fact proprietary applications can use/modify GPL software with no limitations or obligations towards authors. The only requirement is if such applications will be published/distributed it must be done under GPL license. If that's a problem - just ask authors to release a commercial version for such purpose. I think GPL is fair and easy to deal with. Raffaele P. Guidi wrote: > Well, GPL for a library is a dead end. Nobody will use it unless for > open source projects. I would consider any possible choice before > using it. > > Regards, > Raffaele > > On Sat, Jun 20, 2009 at 18:12, paul.degt > wrote: > > As I mentioned the decision was influenced by our legal adviser. > And it > probably can be relicensed as MPL/GPL, but why? > My understanding is that for an end user which wants to use a free > open > source software there should be absolutely no difference between > GPL and > MPL. > But if any one has has any licensing concern with his particular use > case - please contact me with the details, I will do what I can to > help. > > > Brian West wrote: > > I still say why not MPL or at the very least MPL/GPL? > > > > /b > > > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > > > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is > >> defined > >> more clearly from legal perspective, at least from our legal > adviser > >> point of view. > >> > >> Diego Viola wrote: > >> > >>> Because they probably want a stronger copyleft license? > >>> > >>> I prefer the GPL because of that reason. > >>> > >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > >>> > >> > >>> wrote: > >>> > >>> uhm... nice! But why not MPL license (the same as freeswitch)? > >>> > >>> > >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > > >>> > > > >>> >> > wrote: > >>> > >>> http://versafon.com/versafonweb/Software.jsp > >>> > >>> Essentially it's a wrapper around inbound socket interface, > >>> not all > >>> events supported yet, and not all event > parameters/variables. > >>> It's multi > >>> threaded and scaled well in testing. > >>> We offer commercial support and development for > FreeSwitch as > >>> well. > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> > > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > ------------------------------------------------------------------------ > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From raffaele.p.guidi at gmail.com Sat Jun 20 13:49:47 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sat, 20 Jun 2009 22:49:47 +0200 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D436C.6030405@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <4A3D436C.6030405@gmail.com> Message-ID: Right, fair and easy to deal with. And also to NOT deal with - as would any developer working in a commercial company. Again, I would consider any possible choice before using it. Regards, Raffaele On Sat, Jun 20, 2009 at 22:15, paul.degt wrote: > That's rather bold statement and indicates lack of knowledge on the > subject . I would advise you to look at > http://www.fsf.org/licensing/licenses/gpl-faq.html on GPL information. > In fact proprietary applications can use/modify GPL software with no > limitations or obligations towards authors. The only requirement is if > such applications will be published/distributed it must be done under > GPL license. If that's a problem - just ask authors to release a > commercial version for such purpose. > I think GPL is fair and easy to deal with. > > > Raffaele P. Guidi wrote: > > Well, GPL for a library is a dead end. Nobody will use it unless for > > open source projects. I would consider any possible choice before > > using it. > > > > Regards, > > Raffaele > > > > On Sat, Jun 20, 2009 at 18:12, paul.degt > > wrote: > > > > As I mentioned the decision was influenced by our legal adviser. > > And it > > probably can be relicensed as MPL/GPL, but why? > > My understanding is that for an end user which wants to use a free > > open > > source software there should be absolutely no difference between > > GPL and > > MPL. > > But if any one has has any licensing concern with his particular use > > case - please contact me with the details, I will do what I can to > > help. > > > > > > Brian West wrote: > > > I still say why not MPL or at the very least MPL/GPL? > > > > > > /b > > > > > > On Jun 20, 2009, at 9:37 AM, paul.degt wrote: > > > > > > > > >> Yes, that's one of the reasons. Another point is that GPL v.3 is > > >> defined > > >> more clearly from legal perspective, at least from our legal > > adviser > > >> point of view. > > >> > > >> Diego Viola wrote: > > >> > > >>> Because they probably want a stronger copyleft license? > > >>> > > >>> I prefer the GPL because of that reason. > > >>> > > >>> On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi > > >>> > > > > >> > > >>> wrote: > > >>> > > >>> uhm... nice! But why not MPL license (the same as freeswitch)? > > >>> > > >>> > > >>> On Fri, Jun 19, 2009 at 04:06, paul.degt at gmail.com > > > > >>> > > > > > >>> >> > > wrote: > > >>> > > >>> http://versafon.com/versafonweb/Software.jsp > > >>> > > >>> Essentially it's a wrapper around inbound socket > interface, > > >>> not all > > >>> events supported yet, and not all event > > parameters/variables. > > >>> It's multi > > >>> threaded and scaled well in testing. > > >>> We offer commercial support and development for > > FreeSwitch as > > >>> well. > > >>> > > >>> _______________________________________________ > > >>> Freeswitch-users mailing list > > >>> Freeswitch-users at lists.freeswitch.org > > > > >>> > > > > >>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >>> > > >>> _______________________________________________ > > >>> Freeswitch-users mailing list > > >>> Freeswitch-users at lists.freeswitch.org > > > > >>> > > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >>> > > > ------------------------------------------------------------------------ > > >>> > > >>> _______________________________________________ > > >>> Freeswitch-users mailing list > > >>> Freeswitch-users at lists.freeswitch.org > > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >>> > > >>> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/3d05fdc8/attachment-0002.html From anthony.minessale at gmail.com Sat Jun 20 15:10:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 20 Jun 2009 17:10:22 -0500 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <7602CA43-78AB-403A-BEE6-2446A008077C@freeswitch.org> <4A3D0A5F.4080608@gmail.com> <4A3D436C.6030405@gmail.com> Message-ID: <191c3a030906201510t784275b4j17173edc5684a7ab@mail.gmail.com> We don't need to argue licenses the topic is way too subjective. To each his own. Let's concentrate on the code and the cool features we can make in the future. On Jun 20, 2009 4:00 PM, "Raffaele P. Guidi" wrote: Right, fair and easy to deal with. And also to NOT deal with - as would any developer working in a commercial company. Again, I would consider any possible choice before using it. Regards, Raffaele On Sat, Jun 20, 2009 at 22:15, paul.degt wrote: > > That's rather bold statem... _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/b72004b7/attachment-0002.html From steveu at coppice.org Sat Jun 20 17:57:50 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 21 Jun 2009 08:57:50 +0800 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3CF447.3030501@gmail.com> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> Message-ID: <4A3D858E.5050203@coppice.org> paul.degt wrote: > Yes, that's one of the reasons. Another point is that GPL v.3 is defined > more clearly from legal perspective, at least from our legal adviser > point of view. > While the legal status of MPL is widely considered to be vague, is GPL 3 any better? GPL 2 is pretty sound, and has stood the test of time. However a number of large companies have banned their employees from working on anything involving GPL 3 code, because of legal uncertainties, especially with regard to patents. Steve From gcd at i.ph Sat Jun 20 17:58:22 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 08:58:22 +0800 Subject: [Freeswitch-users] FS as a Class 5 switch Message-ID: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> hi everybody, i'm interested to know if anyone employed FS as a local exchange switch. i'm confident FS can handle several calls using RTP by-pass mode. however, i'm more concerned on handling the large dialplan with hundreds (or even a few thousand) exchange prefixes nationwide during call setup. i'd be glad to hear experiences and suggestions esp on the hardware dimensioning. we're talking a small exchange up to about 1,100 lines only, mostly linked to the main exchange via MFC-R2. tks, nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/999823bf/attachment-0002.html From mcampbellsmith at gmail.com Sat Jun 20 18:28:27 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 21 Jun 2009 11:28:27 +1000 Subject: [Freeswitch-users] voicemail problem Message-ID: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> Hi! I have a problem with voicemail in that freeswitch fails to let users leave their message. Something wrong in the config I guess. I see this in the logs: 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message] (en:en) 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) I assume the vm-record_message.PCMU is the file that will be created to record the voicemail. Is that correct and how can I fix this? Thanks! From dyfet at gnutelephony.org Sat Jun 20 18:28:10 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Sat, 20 Jun 2009 21:28:10 -0400 Subject: [Freeswitch-users] MPL and licensing In-Reply-To: <4A3D858E.5050203@coppice.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <4A3D858E.5050203@coppice.org> Message-ID: <4A3D8CAA.1040101@gnutelephony.org> There are no legal uncertainties with respect to patents in GPL v3. You cannot assert them in code you license under it. There was ambiguities in GPL v2 in this respect which some companies liked. I prefer to deal with honest companies rather than those that are anti-social or might choose legal ambush later, so any that feel they cannot accept the greater legal certainty of GPL v3 in this respect are probably companies that I would not choose to have any kind of relationship with anyway ;). I recall there were other technical reasons why some have preferred the MPL, especially over the language of the Lesser GNU General Public License prior to v3. I remember having a lovely discussion about this with Craig Southern a few years back who conceeded that if the language (of the older LGPL) had been corrected for C++ use cases and object oriented practices (inlines, templates, derived classes, etc, all were problems...), he would likely have used it at the time instead of the MPL for OpenH323. Steve Underwood wrote: > paul.degt wrote: >> Yes, that's one of the reasons. Another point is that GPL v.3 is defined >> more clearly from legal perspective, at least from our legal adviser >> point of view. >> > While the legal status of MPL is widely considered to be vague, is GPL 3 > any better? GPL 2 is pretty sound, and has stood the test of time. > However a number of large companies have banned their employees from > working on anything involving GPL 3 code, because of legal > uncertainties, especially with regard to patents. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 177 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/ced168ae/attachment-0002.vcf From msc at freeswitch.org Sat Jun 20 18:40:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 20 Jun 2009 20:40:45 -0500 Subject: [Freeswitch-users] FS as a Class 5 switch In-Reply-To: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> References: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> Message-ID: <87f2f3b90906201840h34bce529oa86267ab50c98c08@mail.gmail.com> Isn't Telco Bridges using it as a switch? -MC On Sat, Jun 20, 2009 at 7:58 PM, Nandy Dagondon wrote: > hi everybody, > > i'm interested to know if anyone employed FS as a local exchange switch. > i'm confident FS can handle several calls using RTP by-pass mode. however, > i'm more concerned on handling the large dialplan with hundreds (or even a > few thousand) exchange prefixes nationwide during call setup. > > i'd be glad to hear experiences and suggestions esp on the hardware > dimensioning. we're talking a small exchange up to about 1,100 lines only, > mostly linked to the main exchange via MFC-R2. > > tks, > nandy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/06e46271/attachment-0002.html From steveu at coppice.org Sat Jun 20 18:52:49 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 21 Jun 2009 09:52:49 +0800 Subject: [Freeswitch-users] MPL and licensing In-Reply-To: <4A3D8CAA.1040101@gnutelephony.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <4A3D858E.5050203@coppice.org> <4A3D8CAA.1040101@gnutelephony.org> Message-ID: <4A3D9271.6030903@coppice.org> David Sugar wrote: > There are no legal uncertainties with respect to patents in GPL v3. You > cannot assert them in code you license under it. There was ambiguities > in GPL v2 in this respect which some companies liked. I prefer to deal > with honest companies rather than those that are anti-social or might > choose legal ambush later, so any that feel they cannot accept the > greater legal certainty of GPL v3 in this respect are probably companies > that I would not choose to have any kind of relationship with anyway ;). > If that is true, why are there some notes on the FSF web site trying to clarify what is not well stated about patents in GPL3 itself? You are fooling yourself if you think lawyers are generally comfortable with GPL3. I'm talking about lawyers who were perfectly happy with GPL2. I know of some cases where large companies have paid outsourced developers to contribute to open source projects, specifically so there are no possible legal ramifications related to their own patent portfolio. That's a really messed up licence. Steve From juanma.v82 at gmail.com Sat Jun 20 19:48:39 2009 From: juanma.v82 at gmail.com (JuanMa) Date: Sat, 20 Jun 2009 23:48:39 -0300 Subject: [Freeswitch-users] Can it do it? In-Reply-To: References: <4982EBF7-7344-4E15-B54D-EEF8DE04E3D6@gmail.com> <92D1B779-43BB-4D24-9E64-914C7E3E005C@gmail.com> <7250FCA5-FF83-4F47-84A0-F473C6FB2326@freeswitch.org> Message-ID: Hi, Saeed thanks for reply my mails. I am only using ONE FS. The Endpoint 1 call to the FS, then it call to SWITCH and the SWITCH call back to the FS and this call to Endpoint 2, my first intention were to use FS in bypass_mode, in this mode Fs only handle SIP signalization and SWITCH (another technology) handle RTP, like the image in the link (http://img26.imageshack.us/img26/1298/fsassbc.png ). Only ONE FS work as SBC (session border controler) or pseudo proxy. But the problem is the FS doesn't negotiate codecs properly and how you wrote late negotiation didn't work for change codec negotiation. I will test your tip, but i cant control the codec negotiation from endpoint who make a first INVITE to FS, only the bridge to de SWICTH and back to FS. Yes, I am using xml_curl. I was testing with event_socket but when FS receive a call and this connect to event_socket, the FS try to park the call and this can't by done because you must control the media. Thanks Regards On 20/06/2009, at 12:06, Saeed Ahmad wrote: > Hi, > > I am not expert, and if i understood you correctly, i am doing it > like that: > > set late negotiation to false (it can be set true if you want to > negotiate it on the fly, but it didn't work for me) > > don't use absolute_codec_string instead use codec_string and put > codec(s) what EP2 is offering, now if A side is sending one of the > codec which are defined for EP2 then it will be used. > > i didn't consider your FS at right side because your SWITCH > definately going to transcode the codec into G729, otherwise above > scenario should be same for both FS. > let me know if it also works for you, i am also using it in proxy > media mode. > are you using xml_curl? > > - Saeed > On Fri, Jun 19, 2009 at 11:28 PM, Mariano de Llano > wrote: > > You are right, it seams that it can not be done. > > In the past I've tried to do something similiar but with no success. > Apperently the documentation is wrong. > > If I have some time I will look at the code and I will give you some > feedback. > > Cheers, > > PS: The transcoding question has nothing to do with your question :) > > On 19/06/2009, at 17:18, JuanMa wrote: > > > I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I > > have another switch who is in charge of it, witch is from another > > technology), I only want to negotiate the codecs in the way that I > > want it. This only seams to work when bypass media or proxy media is > > set to false. Due I need to use it as a SBC(session border > controller) > > or pseudo proxy (I already know that is not intend for it), I need > to > > negotiate the codecs in the FS. In the current thread I've already > > explained what I'm trying to do. If you give me a tip I'm willing to > > make the documentation richer. > > > > Thanks > > Regards > > > > In my architecture the switch who is in charge of transcoding IS > NOT a > > FS. > > > > > > On 19/06/2009, at 16:19, Brian West wrote: > > > >> No right now you can not legally transcode G729 in FreeSWITCH, > >> PERIOD! > >> > >> /b > >> > >> On Jun 19, 2009, at 2:11 PM, JuanMa wrote: > >> > >>> Yes, it can do transcoding. Transcoding isn't the problem to my > >>> architecture, my problem is the codec negotiation between FS and > >>> Endpoints. > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090620/0d191cbe/attachment-0002.html From dave at 3c.co.uk Sat Jun 20 20:50:04 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 21 Jun 2009 06:50:04 +0300 Subject: [Freeswitch-users] FS as a Class 5 switch In-Reply-To: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> References: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> Message-ID: <1245556204.29626.19.camel@dk-d820> Hi Nandy. On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote: > i'm interested to know if anyone employed FS as a local exchange > switch. i'm confident FS can handle several calls using RTP by-pass > mode. however, i'm more concerned on handling the large dialplan with > hundreds (or even a few thousand) exchange prefixes nationwide during > call setup. We have probably ~100k prefixes in our LCR. We don't put these in the dialplan directly; instead, they live in a database and we have an external application which routes calls. FreeSWITCH has mod_lcr which I would imagine will do the same sort of thing; we don't use it because it wasn't around when we started. I'd caution against trying to put thousands of prefixes in the dialplan: I'd guess that matching each call against some thousands of regexes during call setup might get expensive. > i'd be glad to hear experiences and suggestions esp on the hardware > dimensioning. we're talking a small exchange up to about 1,100 lines > only, mostly linked to the main exchange via MFC-R2. That'd depend on the number of concurrent calls you need to budget for - taking it that 1,100 lines implies maybe 1-200 simultaneous calls, then one low-end modern server (Core 2 Duo, etc.) ought to do just fine. Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From gcd at i.ph Sat Jun 20 21:25:58 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 12:25:58 +0800 Subject: [Freeswitch-users] FS as a Class 5 switch In-Reply-To: <1245556204.29626.19.camel@dk-d820> References: <7d0bfd8c0906201758i533beca4k844e8caf7668a253@mail.gmail.com> <1245556204.29626.19.camel@dk-d820> Message-ID: <7d0bfd8c0906202125uec24ad0tc440437fb1f5134e@mail.gmail.com> hi dave, tks for sharing us this info. i don't think we can reach 10k prefixes but your deployment to use external database or mod_lcr is the way to go. re hardware, i think core2 platform would be enough cuz it will be in a rural installation. i'm sure it wont reach 200 simultaneous calls. FS community is really great! tks once again, nandy On Sun, Jun 21, 2009 at 11:50 AM, David Knell wrote: > Hi Nandy. > > On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote: > > i'm interested to know if anyone employed FS as a local exchange > > switch. i'm confident FS can handle several calls using RTP by-pass > > mode. however, i'm more concerned on handling the large dialplan with > > hundreds (or even a few thousand) exchange prefixes nationwide during > > call setup. > > We have probably ~100k prefixes in our LCR. We don't put these in the > dialplan directly; instead, they live in a database and we have an > external application which routes calls. FreeSWITCH has mod_lcr which I > would imagine will do the same sort of thing; we don't use it because it > wasn't around when we started. > > > I'd caution against trying to put thousands of prefixes in the dialplan: > I'd guess that matching each call against some thousands of regexes > during call setup might get expensive. > > > i'd be glad to hear experiences and suggestions esp on the hardware > > dimensioning. we're talking a small exchange up to about 1,100 lines > > only, mostly linked to the main exchange via MFC-R2. > > That'd depend on the number of concurrent calls you need to budget for - > taking it that 1,100 lines implies maybe 1-200 simultaneous calls, then > one low-end modern server (Core 2 Duo, etc.) ought to do just fine. > Cheers -- > > Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/7c61c594/attachment-0002.html From gcd at i.ph Sat Jun 20 22:36:13 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 13:36:13 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem Message-ID: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> hi, i tested the latest SVN build (13884) using the sample configuration files ... no modifications whatsoever. but in sofia external profile, the IP address is my internal address instead of my external IP address. did i miss something here? tks. -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/099963a6/attachment-0002.html From paul.degt at gmail.com Sat Jun 20 22:38:19 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Sun, 21 Jun 2009 01:38:19 -0400 Subject: [Freeswitch-users] Open source Java based inbound event socket library available In-Reply-To: <4A3D858E.5050203@coppice.org> References: <4A3AF28D.4040708@gmail.com> <86a32abc0906200600v133356dft47201934d003580e@mail.gmail.com> <4A3CF447.3030501@gmail.com> <4A3D858E.5050203@coppice.org> Message-ID: <4A3DC74B.4080404@gmail.com> I have Fortune 1000 clients myself, and frankly speaking in real world they don't even care what type of license the free stuff has. Why? Simple. Because 90% of the time these companies buy commercial support and being a commercial customer it's very easy for them to get a commercial version of the software for a reasonable price. But the latter not happening very often - most of the time they never publish their in-house software since it's their competitive advantage - and thus no reason to be concerned about license type. As to "legal uncertainties" - frankly I don't know, my lawyer liked it, knowing how complex the law can be I am sure any type of license free or non-free can potentially bring legal issues - it's imperfect world. Steve Underwood wrote: > paul.degt wrote: > >> Yes, that's one of the reasons. Another point is that GPL v.3 is defined >> more clearly from legal perspective, at least from our legal adviser >> point of view. >> >> > While the legal status of MPL is widely considered to be vague, is GPL 3 > any better? GPL 2 is pretty sound, and has stood the test of time. > However a number of large companies have banned their employees from > working on anything involving GPL 3 code, because of legal > uncertainties, especially with regard to patents. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jason at jasonjgw.net Sat Jun 20 22:45:54 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 21 Jun 2009 15:45:54 +1000 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> Message-ID: <20090621054554.GA24069@jdc.jasonjgw.net> Nandy Dagondon wrote: > hi, > > i tested the latest SVN build (13884) using the sample configuration files > ... no modifications whatsoever. but in sofia external profile, the IP > address is my internal address instead of my external IP address. > > did i miss something here? Try setting ext-sip-ip and ext-rtp-ip in the external profile to stun:stun.freeswitch.org This can alternatively be set using global variables in vars.xml in the supplied configuration. From gcd at i.ph Sat Jun 20 23:20:10 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 14:20:10 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <20090621054554.GA24069@jdc.jasonjgw.net> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> Message-ID: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> the default setting is "auto-nat". i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun:stun.freeswitch.org. result: same problem i tried your suggestion. still the same problem. On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: > Nandy Dagondon wrote: > > hi, > > > > i tested the latest SVN build (13884) using the sample configuration > files > > ... no modifications whatsoever. but in sofia external profile, the IP > > address is my internal address instead of my external IP address. > > > > did i miss something here? > > Try setting ext-sip-ip and ext-rtp-ip in the external profile to > stun:stun.freeswitch.org > > This can alternatively be set using global variables in vars.xml in the > supplied configuration. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/4f2de991/attachment-0002.html From jmesquita at gmail.com Sat Jun 20 23:35:53 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 21 Jun 2009 03:35:53 -0300 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> Message-ID: <5a8712120906202335h1e7a18efo9791a901c6b83c2f@mail.gmail.com> What I would guess is the the STUN lookup failed. Do you have anything on this box that would prevent FS from doing DNS lookup? jmesquita On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and > ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun: > stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. > > > > On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: > >> Nandy Dagondon wrote: >> > hi, >> > >> > i tested the latest SVN build (13884) using the sample configuration >> files >> > ... no modifications whatsoever. but in sofia external profile, the IP >> > address is my internal address instead of my external IP address. >> > >> > did i miss something here? >> >> Try setting ext-sip-ip and ext-rtp-ip in the external profile to >> stun:stun.freeswitch.org >> >> This can alternatively be set using global variables in vars.xml in the >> supplied configuration. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/00aefa7d/attachment-0002.html From gcd at i.ph Sat Jun 20 23:55:08 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 14:55:08 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <5a8712120906202335h1e7a18efo9791a901c6b83c2f@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> <5a8712120906202335h1e7a18efo9791a901c6b83c2f@mail.gmail.com> Message-ID: <7d0bfd8c0906202355r70e57008i78c9ff1ad9df9020@mail.gmail.com> i just come across the Auto NAT feature in the Wiki. i'm testing if my router UPNP works w/ FS. STUN works before i updated to SVN. 2009/6/21 Jo?o Mesquita > What I would guess is the the STUN lookup failed. Do you have anything on > this box that would prevent FS from doing DNS lookup? > > jmesquita > > > On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon wrote: > >> the default setting is "auto-nat". >> >> i changed ext-sip-ip=$${external_sip_ip} and >> ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun: >> stun.freeswitch.org. result: same problem >> >> i tried your suggestion. still the same problem. >> >> >> >> On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: >> >>> Nandy Dagondon wrote: >>> > hi, >>> > >>> > i tested the latest SVN build (13884) using the sample configuration >>> files >>> > ... no modifications whatsoever. but in sofia external profile, the IP >>> > address is my internal address instead of my external IP address. >>> > >>> > did i miss something here? >>> >>> Try setting ext-sip-ip and ext-rtp-ip in the external profile to >>> stun:stun.freeswitch.org >>> >>> This can alternatively be set using global variables in vars.xml in the >>> supplied configuration. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/5e68ab83/attachment-0002.html From gcd at i.ph Sun Jun 21 00:38:25 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 21 Jun 2009 15:38:25 +0800 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> Message-ID: <7d0bfd8c0906210038o17798500s2ae50d6bd9a43f80@mail.gmail.com> it's working now, i mean the Auto NAT feature - after i enabled UPNP feature on my router. it's based on external IP addresses of Ext-SIP-IP and Ext-RTP-IP when performing "sofia status profile [internal|external]" on the cli. however, "sofia status" still shows internal IP address on the external profile. it should display the external IP address instead. On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and > ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun: > stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. > > > > On Sun, Jun 21, 2009 at 1:45 PM, Jason White wrote: > >> Nandy Dagondon wrote: >> > hi, >> > >> > i tested the latest SVN build (13884) using the sample configuration >> files >> > ... no modifications whatsoever. but in sofia external profile, the IP >> > address is my internal address instead of my external IP address. >> > >> > did i miss something here? >> >> Try setting ext-sip-ip and ext-rtp-ip in the external profile to >> stun:stun.freeswitch.org >> >> This can alternatively be set using global variables in vars.xml in the >> supplied configuration. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/0fa7574f/attachment-0002.html From mike at jerris.com Sun Jun 21 01:27:36 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 Jun 2009 04:27:36 -0400 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906210038o17798500s2ae50d6bd9a43f80@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> <7d0bfd8c0906210038o17798500s2ae50d6bd9a43f80@mail.gmail.com> Message-ID: <06F7DAC3-2586-4422-825C-93E1CAE43D0C@jerris.com> It has both the interface bind address and the external address Mike On Jun 21, 2009, at 3:38 AM, Nandy Dagondon wrote: > it's working now, i mean the Auto NAT feature - after i enabled UPNP > feature on my router. it's based on external IP addresses of Ext-SIP- > IP and Ext-RTP-IP when performing "sofia status profile [internal| > external]" on the cli. > > however, "sofia status" still shows internal IP address on the > external profile. it should display the external IP address instead. > > > On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$$ > {external_rtp_ip}. both of them are set in vars.xml as > stun:stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. > > > > On Sun, Jun 21, 2009 at 1:45 PM, Jason White > wrote: > Nandy Dagondon wrote: > > hi, > > > > i tested the latest SVN build (13884) using the sample > configuration files > > ... no modifications whatsoever. but in sofia external profile, > the IP > > address is my internal address instead of my external IP address. > > > > did i miss something here? > > Try setting ext-sip-ip and ext-rtp-ip in the external profile to > stun:stun.freeswitch.org > > This can alternatively be set using global variables in vars.xml in > the > supplied configuration. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/619bdc3f/attachment-0002.html From christian.loeschenkohl at xpirio.com Sun Jun 21 03:05:59 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Sun, 21 Jun 2009 12:05:59 +0200 Subject: [Freeswitch-users] channel variable sip_to_tag Message-ID: <4A3E0607.1060608@xpirio.com> hello do someone know how to get the sip_to_tag from an active call? the sip_from_tag is available as a channel variable but sip_to_tag isn't. i don't know if it is available at call setup, the fist time i see the tag=... in the sip header is the challenge response answer from fs i need this to get my aoc (advice-of-charge) implementation running, this one is based on sip info messages and has to contain the same tag's as the active call. br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From brian at freeswitch.org Sun Jun 21 03:12:03 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 05:12:03 -0500 Subject: [Freeswitch-users] sofia external profile: external IP problem In-Reply-To: <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> References: <7d0bfd8c0906202236x6aabc2cfg80091243fc4bb9bb@mail.gmail.com> <20090621054554.GA24069@jdc.jasonjgw.net> <7d0bfd8c0906202320i48b7b13ckd01440c1f426ba81@mail.gmail.com> Message-ID: <617EB39F-ED75-4969-B79F-4DD7D20E6364@freeswitch.org> Really it shouldn't have changed unless you wiped your configs. The reason it can't work with auto-nat is you're not behind a natpmp or upnp router thus you're going to have to set them manually use stun. You can not use stun for rtp-ip or sip-ip, just for ext-sip-ip and ext- rtp-ip once they are set correctly it should work without a problem. /b On Jun 21, 2009, at 1:20 AM, Nandy Dagondon wrote: > the default setting is "auto-nat". > > i changed ext-sip-ip=$${external_sip_ip} and ext-rtp-ip=$$ > {external_rtp_ip}. both of them are set in vars.xml as > stun:stun.freeswitch.org. result: same problem > > i tried your suggestion. still the same problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/884ae856/attachment-0002.html From jan.kubr at gmail.com Sun Jun 21 04:04:25 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 13:04:25 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT Message-ID: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> We have a SIP gateway behind NAT which I haven't been able to set up to work with Freeswitch. The configuration I thought would work is: This, however, sends REGISTER sip:78.24.13.197;transport=udp SIP/2.0 whereas I need REGISTER sip:11.12.13.10;transport=udp SIP/2.0 When I set "register-proxy" to the public address (78.24.13.197) and comment the "proxy" param out, it successfully registers. Though any other SIP messages are sent to 11.12.13.10 which obviously fails. Basically what I need Freeswitch to do is to send everything to the public address, but put the private one everywhere in the SIP messages. How do I do this? Thanks, Jan From mcampbellsmith at gmail.com Sun Jun 21 04:27:58 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 21 Jun 2009 21:27:58 +1000 Subject: [Freeswitch-users] email core dump Message-ID: <33c87fa30906210427x6833e0adk46d313247ad00ef5@mail.gmail.com> Hi! I am trying to email from 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to 1000 at 192.168.0.20 /bin/cat: write error: Broken pipe sh: line 1: 11975 Done(1) /bin/cat /tmp/mail.124558382500b1 11976 Segmentation fault (core dumped) | exim4 -t myemail at xx.com 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.12455810042c7f] to [myemail at xx.com] I can manually send an email to myself with exim4, but freeswitch fails. Any ideas what I have configured incorrectly? Thanks From jan.kubr at gmail.com Sun Jun 21 04:39:26 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 13:39:26 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> Message-ID: <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> I have found this: http://jira.freeswitch.org/browse/MODENDP-184Thanks to which I know that adding to the profile XML does almost what I need. Is it possible to configure outbound proxy per gateway though? Cheers, Jan On Sun, Jun 21, 2009 at 1:04 PM, Jan Kubr wrote: > We have a SIP gateway behind NAT which I haven't been able to set up > to work with Freeswitch. The configuration I thought would work is: > > > > > > > > > > > This, however, sends > > REGISTER sip:78.24.13.197;transport=udp SIP/2.0 > > whereas I need > > REGISTER sip:11.12.13.10;transport=udp SIP/2.0 > > When I set "register-proxy" to the public address (78.24.13.197) and > comment the "proxy" param out, it successfully registers. Though any > other SIP messages are sent to 11.12.13.10 which obviously fails. > > Basically what I need Freeswitch to do is to send everything to the > public address, but put the private one everywhere in the SIP > messages. How do I do this? > > Thanks, > Jan > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/d7337be1/attachment-0002.html From jan.kubr at gmail.com Sun Jun 21 05:16:11 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 14:16:11 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> Message-ID: <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> Creating a separate sofia profile just for this gateway definitely works, just wondering whether there is a cleaner solution. The register-proxy params seems to do something very similar.. On Sun, Jun 21, 2009 at 1:39 PM, Jan Kubr wrote: > I have found this: http://jira.freeswitch.org/browse/MODENDP-184 Thanks to > which I know that adding > > > > to the profile XML does almost what I need. Is it possible to configure > outbound proxy per gateway though? > > Cheers, > Jan > > On Sun, Jun 21, 2009 at 1:04 PM, Jan Kubr wrote: > >> We have a SIP gateway behind NAT which I haven't been able to set up >> to work with Freeswitch. The configuration I thought would work is: >> >> >> >> >> >> >> >> >> >> >> This, however, sends >> >> REGISTER sip:78.24.13.197;transport=udp SIP/2.0 >> >> whereas I need >> >> REGISTER sip:11.12.13.10;transport=udp SIP/2.0 >> >> When I set "register-proxy" to the public address (78.24.13.197) and >> comment the "proxy" param out, it successfully registers. Though any >> other SIP messages are sent to 11.12.13.10 which obviously fails. >> >> Basically what I need Freeswitch to do is to send everything to the >> public address, but put the private one everywhere in the SIP >> messages. How do I do this? >> >> Thanks, >> Jan >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/dfeea9c2/attachment-0002.html From brian at freeswitch.org Sun Jun 21 05:17:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 07:17:21 -0500 Subject: [Freeswitch-users] email core dump In-Reply-To: <33c87fa30906210427x6833e0adk46d313247ad00ef5@mail.gmail.com> References: <33c87fa30906210427x6833e0adk46d313247ad00ef5@mail.gmail.com> Message-ID: <7C7A8ED9-ECED-4100-87F6-0875C054EC64@freeswitch.org> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings /b On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: > Hi! > > I am trying to email from > 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore > original codec. > 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to > 1000 at 192.168.0.20 > /bin/cat: write error: Broken pipe > sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. > 124558382500b1 > 11976 Segmentation fault (core dumped) | exim4 -t myemail at xx.com > 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file > [/tmp/mail.12455810042c7f] to [myemail at xx.com] > > I can manually send an email to myself with exim4, but freeswitch > fails. > > Any ideas what I have configured incorrectly? > > Thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sun Jun 21 05:19:22 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 07:19:22 -0500 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> Message-ID: They usually will not auth on the RURI... I recommend you set the from- domain on your gateway... I think thats really what you need. /b On Jun 21, 2009, at 6:39 AM, Jan Kubr wrote: > I have found this: http://jira.freeswitch.org/browse/MODENDP-184 > Thanks to which I know that adding > > > > to the profile XML does almost what I need. Is it possible to > configure outbound proxy per gateway though? > > Cheers, > Jan > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/4050689d/attachment-0002.html From brian at freeswitch.org Sun Jun 21 05:25:08 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 21 Jun 2009 07:25:08 -0500 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> Message-ID: <981C2E3E-3BF6-4629-A0EE-FFB47B792104@freeswitch.org> nobody authenticates on the request URI... you're focusing on the wrong thing... you'll need from-domain and/or from-user I suspect. /b On Jun 21, 2009, at 7:16 AM, Jan Kubr wrote: > Creating a separate sofia profile just for this gateway definitely > works, just wondering whether there is a cleaner solution. The > register-proxy params seems to do something very similar.. From larclap at yahoo.com Sun Jun 21 10:36:19 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sun, 21 Jun 2009 10:36:19 -0700 Subject: [Freeswitch-users] svn update error Message-ID: <006f01c9f296$c99f0d30$5cdd2790$@com> I am currently running 13723 and want to get current. When I issue the "svn update" command, the following error appears: A src/mod/asr_tts/mod_unimrcp A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.2008.vcproj A src/mod/asr_tts/mod_unimrcp/unimrcp.vsprops A src/mod/asr_tts/mod_unimrcp/Makefile.am A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c U src/mod/asr_tts/mod_pocketsphinx/mod_pocketsphinx.c U src/mod/event_handlers/mod_event_multicast/mod_event_multicast.c svn: Failed to add file 'src/mod/event_handlers/mod_event_multicast/Makefile': an unversioned file of the same name already exists What do I need to do? Should I delete the Makefile file? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/3f8a1a89/attachment-0002.html From mrene_lists at avgs.ca Sun Jun 21 10:38:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 21 Jun 2009 13:38:17 -0400 Subject: [Freeswitch-users] svn update error In-Reply-To: <006f01c9f296$c99f0d30$5cdd2790$@com> References: <006f01c9f296$c99f0d30$5cdd2790$@com> Message-ID: yeah Math On 21-Jun-09, at 1:36 PM, Lars Zeb wrote: > I am currently running 13723 and want to get current. When I issue > the ?svn update? command, the following error appears: > > A src/mod/asr_tts/mod_unimrcp > A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.2008.vcproj > A src/mod/asr_tts/mod_unimrcp/unimrcp.vsprops > A src/mod/asr_tts/mod_unimrcp/Makefile.am > A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c > U src/mod/asr_tts/mod_pocketsphinx/mod_pocketsphinx.c > U src/mod/event_handlers/mod_event_multicast/mod_event_multicast.c > svn: Failed to add file 'src/mod/event_handlers/mod_event_multicast/ > Makefile': an unversioned file of the same name already exists > > What do I need to do? Should I delete the Makefile file? > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/163338ed/attachment-0002.html From jan.kubr at gmail.com Sun Jun 21 10:44:46 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Sun, 21 Jun 2009 19:44:46 +0200 Subject: [Freeswitch-users] SIP gateway behind NAT In-Reply-To: <981C2E3E-3BF6-4629-A0EE-FFB47B792104@freeswitch.org> References: <698401620906210404m455fd852xd2b49ab3f6740f13@mail.gmail.com> <698401620906210439w3e9f4c41wb9bae7ee0de7bc4d@mail.gmail.com> <698401620906210516s22d9d236t8848fdedcaf9ebf3@mail.gmail.com> <981C2E3E-3BF6-4629-A0EE-FFB47B792104@freeswitch.org> Message-ID: <698401620906211044u7f989321k769481e1bf3943c7@mail.gmail.com> Well when I duplicate the external profile, add to it and make the gateway configuration this: Then it does what I need. I was just curious if I can achieve the same without creating a new profile. Jan On Sun, Jun 21, 2009 at 2:25 PM, Brian West wrote: > nobody authenticates on the request URI... you're focusing on the > wrong thing... you'll need from-domain and/or from-user I suspect. > > /b > > On Jun 21, 2009, at 7:16 AM, Jan Kubr wrote: > > > Creating a separate sofia profile just for this gateway definitely > > works, just wondering whether there is a cleaner solution. The > > register-proxy params seems to do something very similar.. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/b10f3398/attachment-0002.html From william.suffill at gmail.com Sun Jun 21 11:44:15 2009 From: william.suffill at gmail.com (William Suffill) Date: Sun, 21 Jun 2009 14:44:15 -0400 Subject: [Freeswitch-users] svn update error In-Reply-To: References: <006f01c9f296$c99f0d30$5cdd2790$@com> Message-ID: <6b65470d0906211144l1bf9f6feyaee0854fad28eece@mail.gmail.com> Seems there was some major changes depending on what revision you are trying to update from. A fresh SVN checkout might be the fastest fix and what I did here once I couldn't update very old checkouts. Technically if you slowly updated it from revision -> revision until it was current that too should work. Not worth the hassle IMHO tho. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/e23693b4/attachment-0002.html From drago at windstream.net Sun Jun 21 12:51:31 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 21 Jun 2009 15:51:31 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: <000001c9ed45$72058320$56108960$@net> References: <000001c9ed45$72058320$56108960$@net> Message-ID: <00cd01c9f2a9$ad1ec700$075c5500$@net> After days of running around clueless, I have no other option but to ask the community for help one last time. Here is what happens: 1. FS sends INVITE 2. Exchange answers with "302 Moved Temporarily" 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: "MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The 'From' field's address to be in the format ''; otherwise, MS Exchange drops the call." I don't know if this is the only problem. However, I see exactly this behavior: "PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established." After "302 (Moved Temporarily", FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/3dbb2445/attachment-0002.html From mike at jerris.com Sun Jun 21 13:31:04 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 Jun 2009 16:31:04 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: <00cd01c9f2a9$ad1ec700$075c5500$@net> References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: > After days of running around clueless, I have no other option but to > ask the community for help one last time? > > > > Here is what happens: > > > > 1. FS sends INVITE > > 2. Exchange answers with ?302 Moved Temporarily? > > 3. FS bombs and closes > > > > > > Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/ > 2.0 > > Message Header > > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > Max-Forwards: 68 > > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > To: > > SIP to address: sip:4783874764 at 10.0.0.71 > > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > CSeq: 116688296 INVITE > > Contact: > > Contact Binding: +um.gmc.cc.ga.us at 10.8.4.3:5060;transport=tcp> > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Privacy: none > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 236 > > P-Asserted-Identity: "MILLEDGEVL GA" > > > Message Body > > > > > > > > Status-Line: SIP/2.0 100 Trying > > Message Header > > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > TO: > > SIP to address: sip:4783874764 at 10.0.0.71 > > CSEQ: 116688296 INVITE > > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > CONTENT-LENGTH: 0 > > > > > > Status-Line: SIP/2.0 302 Moved Temporarily > > Message Header > > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > TO: ;tag=27df6afe0 > > SIP to address: sip:4783874764 at 10.0.0.71 > > SIP tag: 27df6afe0 > > CSEQ: 116688296 INVITE > > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > CONTACT: > > Contact Binding: 4783874764 at 10.0.0.71:5065;transport=tcp> > > CONTENT-LENGTH: 0 > > SERVER: RTCC/3.0.0.0 > > > > > > Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 > > Message Header > > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > > Max-Forwards: 68 > > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > > SIP Display info: "MILLEDGEVL GA" > > SIP from address: sip:4782512197 at 10.8.4.3 > > SIP tag: Qr4j0XX18XD1m > > To: ;tag=27df6afe0 > > SIP to address: sip:4783874764 at 10.0.0.71 > > SIP tag: 27df6afe0 > > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > > CSeq: 116688296 ACK > > Content-Length: 0 > > > > > > Here is the FS log beginning the the processing of the call: > > > > 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel > sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937- > b321c0d87414] > > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/ > external/4782512197 at 209.249.3.59 entering state [received][100] > > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=NXT02 19785 8060 IN IP4 209.249.3.59 > > s=sip call > > c=IN IP4 209.249.3.60 > > t=0 0 > > m=audio 36292 RTP/AVP 0 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec > Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/ > external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples > > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf > payload to 101 > > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW > > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 > (sofia/external/4782512197 at 209.249.3.59) State NEW > > 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_NEW -> CS_INIT > > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 > (sofia/external/4782512197 at 209.249.3.59) State INIT > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 > SOFIA INIT > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_INIT -> CS_ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 > (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change > CS_ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 > (sofia/external/4782512197 at 209.249.3.59) State ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 > SOFIA ROUTING > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 > sofia/external/4782512197 at 209.249.3.59 Standard ROUTING > > 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing > MILLEDGEVL GA->4783874190 in context public > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >unloop] continue=false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >outside_call] continue=true > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition > [outside_call] > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > set(outside_call=true) > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >call_debug] continue=true > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [call_debug] ${call_debug}(true) =~ /^true$/ break=never > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >public_extensions] continue=false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9]) > $/ break=on-false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >Local_UM] continue=false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on- > false > > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 > (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> > CS_EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 > (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 > (sofia/external/4782512197 at 209.249.3.59) Running State Change > CS_EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 > (sofia/external/4782512197 at 209.249.3.59) State EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 > SOFIA EXECUTE > > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 > sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE > > EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) > > 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 > SET [outside_call]=[true] > > EXECUTE sofia/external/4782512197 at 209.249.3.59 info() > > 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: > > Channel-State: [CS_EXECUTE] > > Channel-State-Number: [4] > > Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > > Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > > Call-Direction: [inbound] > > Presence-Call-Direction: [inbound] > > Answer-State: [ringing] > > Channel-Read-Codec-Name: [PCMU] > > Channel-Read-Codec-Rate: [8000] > > Channel-Write-Codec-Name: [PCMU] > > Channel-Write-Codec-Rate: [8000] > > Caller-Username: [4782512197] > > Caller-Dialplan: [XML] > > Caller-Caller-ID-Name: [MILLEDGEVL GA] > > Caller-Caller-ID-Number: [4782512197] > > Caller-Network-Addr: [209.249.3.59] > > Caller-Destination-Number: [4783874190] > > Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > > Caller-Source: [mod_sofia] > > Caller-Context: [public] > > Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > > Caller-Profile-Index: [1] > > Caller-Profile-Created-Time: [1245613758636018] > > Caller-Channel-Created-Time: [1245613758636018] > > Caller-Channel-Answered-Time: [0] > > Caller-Channel-Progress-Time: [0] > > Caller-Channel-Progress-Media-Time: [0] > > Caller-Channel-Hangup-Time: [0] > > Caller-Channel-Transfer-Time: [0] > > Caller-Screen-Bit: [true] > > Caller-Privacy-Hide-Name: [false] > > Caller-Privacy-Hide-Number: [false] > > variable_sip_received_ip: [209.249.3.59] > > variable_sip_received_port: [5060] > > variable_sip_via_protocol: [udp] > > variable_sip_from_user: [4782512197] > > variable_sip_from_uri: [4782512197 at 209.249.3.59] > > variable_sip_from_host: [209.249.3.59] > > variable_sip_from_user_stripped: [4782512197] > > variable_sip_from_tag: [3454602382-411732] > > variable_sofia_profile_name: [external] > > variable_sip_cid_type: [pid] > > variable_sip_req_user: [gw+Broadvox] > > variable_sip_req_port: [5080] > > variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] > > variable_sip_req_host: [71.29.0.61] > > variable_sip_to_user: [4783874764] > > variable_sip_to_port: [5060] > > variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] > > variable_sip_to_host: [209.249.3.56] > > variable_sip_contact_user: [4782512197] > > variable_sip_contact_port: [5060] > > variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] > > variable_sip_contact_host: [209.249.3.59] > > variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] > > variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] > > variable_sip_via_host: [209.249.3.59] > > variable_sip_via_port: [5060] > > variable_max_forwards: [69] > > variable_sip_call_info: [ 209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"] > > variable_sip_gateway: [Broadvox] > > variable_switch_r_sdp: [v=0 > > o=NXT02 19785 8060 IN IP4 209.249.3.59 > > s=sip call > > c=IN IP4 209.249.3.60 > > t=0 0 > > m=audio 36292 RTP/AVP 0 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > ] > > variable_remote_media_ip: [209.249.3.60] > > variable_remote_media_port: [36292] > > variable_read_codec: [PCMU] > > variable_read_rate: [8000] > > variable_write_codec: [PCMU] > > variable_write_rate: [8000] > > variable_endpoint_disposition: [RECEIVED] > > variable_outside_call: [true] > > variable_current_application: [info] > > > > > > EXECUTE sofia/external/4782512197 at 209.249.3.59 > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > > 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 > variable string 0 = [absolute_codec_string=PCMA] > > 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] > > 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/ > 4783874764) State Change CS_NEW -> CS_INIT > > 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_INIT > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT > > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 > SOFIA INIT > > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 4783874764) State Change CS_INIT -> CS_ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT going to sleep > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 > SOFIA ROUTING > > 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/ > internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING going to sleep > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA > > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 > (sofia/internal/4783874764) State CONSUME_MEDIA > > 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/ > internal/4783874764 entering state [calling][0] > > > > > > Will trade my first born for little help J > > > > Drago > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Drago Totev > Sent: Sunday, June 14, 2009 7:11 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 > > > > Hello everyone, > > > > I am trying to set a test environment with FS and Exchange 2007. > > > > Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM > and it does not seem to work. Thus far I use the default settings, > one ATA registered and confirmed to work. > > > > According external source: > > > > ?MS Exchange does not understand 'From' fields with domain name in t > he address (example: field containing addre > ss).The ?From? field?s address to be in the format > ??; otherwise, MS Exchange drops the call.? > > > > I don?t know if this is the only problem? However, I see exactly thi > s behavior: > > > > ?PBX initiates call to MS Exchange by sending a SIP INVITE message t > o the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Tempora > rily) response to PBX asking to repeat INVITE to a different port (5 > 065 for example). After PBX repeats the INVITE sending, the call is > established.? > > > > After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without > any record in the log. > > > > Can someone help with working configuration, please? Exchange 2007 > UM role Version: 08.01.0359.002 > > > > Thanks in advance. > > > > Drago > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/85be5cc9/attachment-0002.html From drago at windstream.net Sun Jun 21 14:21:07 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 21 Jun 2009 17:21:07 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: <00de01c9f2b6$31d14de0$9573e9a0$@net> Wiki does not have instruction how to do so on Windows (I should have mention this in the beginning ? sorry about that). Can I do it anyway? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Sunday, June 21, 2009 4:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: After days of running around clueless, I have no other option but to ask the community for help one last time? Here is what happens: 1. FS sends INVITE 2. Exchange answers with ?302 Moved Temporarily? 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: ?MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The ?From? field?s address to be in the format ??; otherwise, MS Exchange drops the call.? I don?t know if this is the only problem? However, I see exactly this behavior: ?PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established.? After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/3c4be4b6/attachment-0002.html From drago at windstream.net Sun Jun 21 14:23:45 2009 From: drago at windstream.net (Drago Totev) Date: Sun, 21 Jun 2009 17:23:45 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: <00e301c9f2b6$8f710c10$ae532430$@net> The Windows Application Log shows this, however: 1331177817 1 APPCRASH Not available 0 FreeSwitch.exe 0.0.0.0 4a316e86 MSVCR90.dll 9.0.30729.4918 49d43da7 c0000005 0003b590 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Sunday, June 21, 2009 4:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: After days of running around clueless, I have no other option but to ask the community for help one last time? Here is what happens: 1. FS sends INVITE 2. Exchange answers with ?302 Moved Temporarily? 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: ?MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The ?From? field?s address to be in the format ??; otherwise, MS Exchange drops the call.? I don?t know if this is the only problem? However, I see exactly this behavior: ?PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established.? After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/ea5968f1/attachment-0002.html From larclap at yahoo.com Sun Jun 21 17:42:40 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sun, 21 Jun 2009 17:42:40 -0700 Subject: [Freeswitch-users] event_add_head warning message on cosole Message-ID: <00bf01c9f2d2$594f1a20$0bed4e60$@com> I just upgraded to version 13886. On the console the following messages appear every few minutes. I've looked at the code but it's way over my head. Why is it displaying? How can I turn it off? 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'toll_allow' = 'domestic,international,local' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'accountcode' = '1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'user_context' = 'default' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'effective_caller_id_name' = 'Extension 1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'effective_caller_id_number' = '1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'outbound_caller_id_number' = '3235551212' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'callgroup' = 'techsupport' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'record_stereo' = 'true' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'default_gateway' = 'example.com' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'default_areacode' = '323' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'transfer_fallback_extension' = 'operator' Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090621/52305a54/attachment-0002.html From JCasale at activenetwerx.com Sun Jun 21 18:25:56 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 01:25:56 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk Message-ID: I attempted to build rpm's from the included spec file using a non-root user build environment. The steps I used are as follows: 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as expected by the spec: svn co http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4 tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ 3. Copy spec to SPECS directory: 4. Pull in Source (Source0 doesn't exist yet, I make it above): for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done 5. Check spec in svn copy for deps: yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' freeswitch.spec) 6.Build rpm: rpmbuild -ba freeswitch.spec After some time, near the end I see various issues like the following: making install mod_speex installing mod_speex.so quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/freeswitch-1.0.4/libfreeswitch.la' has not been installed in `/opt/freeswitch/lib' It also seems to download everything it would normally again, then fails with several errors like the following: RPM build errors: File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/opt/freeswitch/mod/ozmod_wanpipe.so* Installed (but unpackaged) file(s) found: /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml . . . After which no rpm's are built? Anyone know what tricks are still needed with the spec from svn? Thanks! jlc From mike at jerris.com Sun Jun 21 18:45:01 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 21 Jun 2009 21:45:01 -0400 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: Message-ID: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> It's failing to build the core library. There should be some warning before it tried to build the modules in the log. On Jun 21, 2009, at 9:25 PM, "Joseph L. Casale" wrote: > I attempted to build rpm's from the included spec file using a non- > root user > build environment. The steps I used are as follows: > > 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS > 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as > expected by the spec: > svn co http://svn.freeswitch.org/svn/freeswitch/trunk > freeswitch-1.0.4 > tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ > 3. Copy spec to SPECS directory: > 4. Pull in Source (Source0 doesn't exist yet, I make it above): > for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' > freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done > 5. Check spec in svn copy for deps: > yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' > freeswitch.spec) > 6.Build rpm: > rpmbuild -ba freeswitch.spec > > After some time, near the end I see various issues like the following: > > making install mod_speex > installing mod_speex.so > quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/ > freeswitch-1.0.4/libfreeswitch.la' > has not been installed in `/opt/freeswitch/lib' > > > It also seems to download everything it would normally again, then > fails with > several errors like the following: > > RPM build errors: > File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/ > opt/freeswitch/mod/ozmod_wanpipe.so* > Installed (but unpackaged) file(s) found: > /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml > /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml > /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml > /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml > . > . > . > > After which no rpm's are built? Anyone know what tricks are still > needed with > the spec from svn? > > Thanks! > jlc > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Sun Jun 21 21:08:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 22 Jun 2009 00:08:07 -0400 Subject: [Freeswitch-users] event_add_head warning message on cosole In-Reply-To: <00bf01c9f2d2$594f1a20$0bed4e60$@com> References: <00bf01c9f2d2$594f1a20$0bed4e60$@com> Message-ID: <802DCA59-126E-4D33-A910-C95C7644CACB@avgs.ca> 13887 created by brian on 21 June 2009, 21:52:31 -0500 (75 minutes ago) (patch) move this to debug and profile->debug so that its not on unless you enable the profile debug also. That should fix it, update again :) Math On 21-Jun-09, at 8:42 PM, Lars Zeb wrote: > I just upgraded to version 13886. On the console the following > messages appear every few minutes. I?ve looked at the code but it?s > way over my head. > > Why is it displaying? How can I turn it off? > > 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 > event_add_header -> 'toll_allow' = 'domestic,international,local' > 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 > event_add_header -> 'accountcode' = '1000' > 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 > event_add_header -> 'user_context' = 'default' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/61fa3c57/attachment-0002.html From brian at freeswitch.org Sun Jun 21 22:51:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 01:51:02 -0400 Subject: [Freeswitch-users] event_add_head warning message on cosole In-Reply-To: <802DCA59-126E-4D33-A910-C95C7644CACB@avgs.ca> References: <00bf01c9f2d2$594f1a20$0bed4e60$@com> <802DCA59-126E-4D33-A910-C95C7644CACB@avgs.ca> Message-ID: <07B4A4D4-06E4-4A88-8E96-204C971D246A@freeswitch.org> it was actually tony but who cares! ;) /b On Jun 22, 2009, at 12:08 AM, Mathieu Rene wrote: > 13887 created by brian on 21 June 2009, 21:52:31 -0500 (75 minutes > ago) (patch) move this to debug and profile->debug so that its not > on unless you enable the profile debug also. > > That should fix it, update again :) > > Math > > On 21-Jun-09, at 8:42 PM, Lars Zeb wrote: > >> I just upgraded to version 13886. On the console the following >> messages appear every few minutes. I?ve looked at the code but it?s >> way over my head. >> >> Why is it displaying? How can I turn it off? >> >> 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 >> event_add_header -> 'toll_allow' = 'domestic,international,local' >> 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 >> event_add_header -> 'accountcode' = '1000' >> 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 >> event_add_header -> 'user_context' = 'default' > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b8bdebec/attachment-0002.html From d at d-man.org Mon Jun 22 00:31:57 2009 From: d at d-man.org (Darren Schreiber) Date: Mon, 22 Jun 2009 00:31:57 -0700 Subject: [Freeswitch-users] Question about bridging calls to a specific URI via a specific profile Message-ID: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> Hello, I was wondering, I am bridging a call to a specific URI as follows: EXECUTE sofia/internal/+17209460000@ 2.3.4.5 bridge(sofia/internal/3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip at 3As%403.55.66.180%3A7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/URI at domain that FS will use the specified SIP profile to try and connect a call to the URI at domain specified. Since the full URI at domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:XXX at domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/5b6f5687/attachment-0002.html From d at d-man.org Mon Jun 22 01:06:16 2009 From: d at d-man.org (Darren Schreiber) Date: Mon, 22 Jun 2009 01:06:16 -0700 Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile In-Reply-To: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> References: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> Message-ID: <6EA6ABD53A184F1F90CF51F883729E8E@test> Ignore this thread. Apparently I was stripping sip: from the prefix. I guess you have to specify sip: before utilizing fs_nat and fs_path variables. My bad. _____ From: Darren Schreiber [mailto:d at d-man.org] Sent: Monday, June 22, 2009 12:32 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile Hello, I was wondering, I am bridging a call to a specific URI as follows: EXECUTE sofia/internal/+17209460000@ 2.3.4.5 bridge(sofia/internal/3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223232 at 3.55.66.180;fs_nat=yes;fs_path=sip at 3As%403.55.66.180%3A7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/URI at domain that FS will use the specified SIP profile to try and connect a call to the URI at domain specified. Since the full URI at domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:XXX at domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b78961e6/attachment-0002.html From dome at tel.co.th Mon Jun 22 01:10:28 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 22 Jun 2009 15:10:28 +0700 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> Message-ID: <8ccbff060906220110y2b9431b5g797d4122152891d9@mail.gmail.com> default_language still don't work wirh say but sound_prefix work fine. example my javascript ------------------------------- session.execute("set", "sound_prefix=/opt/freeswitch/sounds/th"); session.execute("say","th number pronounced 1346523"); session.execute("say","th number pronounced 21"); session.execute("say","th number pronounced 11"); session.execute("say","th number pronounced 101"); How to check in mod_say_th back to freeswotch ? Dome C. 2009/6/3 Brian West : > You'll need to set the variable?default_language > /b > On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote: > > Dear sir, > ????????i create mod_say_th for Thai language. i found some problem > about sound-path. > I have config th.xml in conf/lang/th/ > tts-engine="cepstral" tts-voice="callie"> > ... > > when i try > > Freeswitch still looking sounf file ?in ?/sounds/en/us/callie ?(en > sound-path) > > Someone help me please > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Mon Jun 22 01:38:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 01:38:41 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED Message-ID: <24143545.post@talk.nabble.com> Hi, API CALL [originate sofia/external/1001 at 116.50.456.212] -ERR SERVICE_NOT_IMPLEMENTED I receiving this error i dont know y? Can u help mo on this? I dialing a mobile number on this sometimes it works... Sometimes it destroys the call [CALL_DESTROY] Thanks -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24143545.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 22 01:39:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 01:39:59 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED Message-ID: <24143545.post@talk.nabble.com> Hi, API CALL [originate sofia/external/1001 at 116.50.456.212 1001] output: -ERR SERVICE_NOT_IMPLEMENTED I receiving this error i dont know y? Can u help mo on this? I dialing a mobile number on this sometimes it works... Sometimes it destroys the call [CALL_DESTROY] I already installed zrtp softphone connecting to mobile phones... Thanks -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24143545.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From maarten at nalawo.com Mon Jun 22 02:49:09 2009 From: maarten at nalawo.com (MaartenDM) Date: Mon, 22 Jun 2009 02:49:09 -0700 (PDT) Subject: [Freeswitch-users] Nibblebill heartbeat on B-leg Message-ID: <24144481.post@talk.nabble.com> Hello, I am using Nibblebill to bill bridged calls initiated via API. The problem is that the billing on the B-leg is only done when the call is terminated and not at heartbeat. For the a-leg it is working. I added the global_heartbeat variable at the b-leg but without success. I use now: originate {ignore_early_media=true,nibble_account=100,nibble_rate=0.02}sofia/external/12345 at serverip &bridge({global_heartbeat=50,nibble_account=100,nibble_rate=0.02}sofia/external/45678 at serverip) thx, MdM -- View this message in context: http://www.nabble.com/Nibblebill-heartbeat-on-B-leg-tp24144481p24144481.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 22 06:22:24 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 09:22:24 -0400 Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile In-Reply-To: <6EA6ABD53A184F1F90CF51F883729E8E@test> References: <9F412D9DB8B947D9A40E40A5DA48ECDB@test> <6EA6ABD53A184F1F90CF51F883729E8E@test> Message-ID: bingo! :P /b On Jun 22, 2009, at 4:06 AM, Darren Schreiber wrote: > Ignore this thread. Apparently I was stripping sip: from the prefix. > I guess you have to specify sip: before utilizing fs_nat and fs_path > variables. > > My bad. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/a0a128e8/attachment-0002.html From brian at freeswitch.org Mon Jun 22 06:23:13 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 09:23:13 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED In-Reply-To: <24143545.post@talk.nabble.com> References: <24143545.post@talk.nabble.com> Message-ID: <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> I'm going to guess you're calling a registered user? If so replace the @ with % /b On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: > > Hi, > > API CALL [originate sofia/external/1001 at 116.50.456.212] > -ERR SERVICE_NOT_IMPLEMENTED > > I receiving this error i dont know y? Can u help mo on this? > > I dialing a mobile number on this sometimes it works... Sometimes it > destroys the call [CALL_DESTROY] > > > Thanks Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/4d946fc7/attachment-0002.html From brian at freeswitch.org Mon Jun 22 06:23:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 09:23:40 -0400 Subject: [Freeswitch-users] How to change sound-path when switch language In-Reply-To: <8ccbff060906220110y2b9431b5g797d4122152891d9@mail.gmail.com> References: <8ccbff060906021242q29de322fg23fc3068736995ef@mail.gmail.com> <42FF17F6-916E-44EF-8D57-FE2F58F1423B@freeswitch.org> <8ccbff060906220110y2b9431b5g797d4122152891d9@mail.gmail.com> Message-ID: <96FA3310-7670-4540-8F0F-CE1EB2F5D1D1@freeswitch.org> Please open a jira about this. /b On Jun 22, 2009, at 4:10 AM, Dome Charoenyost wrote: > default_language still don't work wirh say > but sound_prefix work fine. > example my javascript > ------------------------------- > session.execute("set", "sound_prefix=/opt/freeswitch/sounds/th"); > session.execute("say","th number pronounced 1346523"); > session.execute("say","th number pronounced 21"); > session.execute("say","th number pronounced 11"); > session.execute("say","th number pronounced 101"); > > > How to check in mod_say_th back to freeswotch ? > > Dome C. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/06c3f15e/attachment-0002.html From JCasale at activenetwerx.com Mon Jun 22 07:36:30 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 14:36:30 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> Message-ID: >It's failing to build the core library. There should be some warning >before it tried to build the modules in the log. You'll have to bear with me, I am not sure exactly what part of the build that is, I see this in the log: +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +----------------------------------------------+ followed by this: +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + ------------------------------------ + and finally it ends with: Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/freeswitch-1.0.4-1-root-rpmbuilder RPM build errors: Which doesn't help:) Thanks! jlc From brian at freeswitch.org Mon Jun 22 07:45:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 10:45:02 -0400 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> Message-ID: <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> Scroll up and look for the real error. /b On Jun 22, 2009, at 10:36 AM, Joseph L. Casale wrote: > You'll have to bear with me, I am not sure exactly what part of the > build > that is, I see this in the log: > > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/bin/make install + > +----------------------------------------------+ > > followed by this: > > +-------- FreeSWITCH install Complete ----------+ > + FreeSWITCH has been successfully installed. + > + + > + Install sounds: + > + (uhd-sounds includes hd-sounds, sounds) + > + (hd-sounds includes sounds) + > + ------------------------------------ + > > and finally it ends with: > > Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/ > freeswitch-1.0.4-1-root-rpmbuilder > > > RPM build errors: > > Which doesn't help:) > Thanks! > jlc Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/7886118d/attachment-0002.html From max.bridgewater at gmail.com Mon Jun 22 08:57:18 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 22 Jun 2009 11:57:18 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Message-ID: Hi Mike, Unfortunately this doesn't seem to solve my problem. Here is my extension again: I've copied it now under: /user/local/freeswitch/conf/dialplan/default /user/local/freeswitch/conf/dialplan/public The different dial strings i tried: "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 & park()" "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67" "{origination_caller_id_number=120003}sofia/internal/242424" "{origination_caller_id_number=120003}sofia/internal/242424%192.168.50.62" My goal: have the call captured by the above extension and redirected to a server socket running at 192.168.50.67:10000. Any thought? Max. On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater wrote: > I don't have my settings to try it right now. Still i have a question. If > it's the way you describe it, why wouldn't sofia/extenal/foo at bar solve the > problem? I think i even copied the extension both to the default directory. > But i will confirm and let you know. > > Max. > > > On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins wrote: > >> Now I feel stupid because I didn't read your original post closely >> enough. >> >> You've defined your "mysocket" extension in the "public" context but when >> you do an origination with sofia/internal/foo at bar it will use the >> "default" context. I think the quickest way to handle this is to create a >> copy of your mysocket.xml file and put it in conf/dialplan/default/ and be >> done with it. >> >> -MC >> >> >> On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Mike, >>> >>> Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML >>> to me though. >>> >>> Max. >>> >>> >>> On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: >>> >>>> Max, >>>> that pastebin failed miserably as none of the xml shows up. can you try >>>> again or use our pastebin.freeswitch.org site? >>>> -MC >>>> >>>> >>>> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >>>> max.bridgewater at gmail.com> wrote: >>>> >>>>> Hi Mike, >>>>> >>>>> It's pasted here: http://pastebin.ca/1466521 >>>>> >>>>> Thanks, >>>>> Max. >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote: >>>>> >>>>>> Can you turn on debugging (F8) and capture all the output after your >>>>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>>>> -MC >>>>>> >>>>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>>>> max.bridgewater at gmail.com> wrote: >>>>>> >>>>>>> Any help our there? >>>>>>> >>>>>>> I'm still trying to get this piece working. Essentially what i wan to >>>>>>> do is, when a call comes in (from registered devices as well as unregistered >>>>>>> devices), notify the my server socket. Somehow it's not working. The change >>>>>>> i made compared to the standard Freeswitch settings are the following: >>>>>>> >>>>>>> 1) Added following extension that in >>>>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2) Changed file: >>>>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> I noticed that with this extension, all calls received from external >>>>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>>>> But calls from registered devices and initiated using the socket interface >>>>>>> are not forwarded. Is there something that need to be changed in the >>>>>>> profiles? >>>>>>> >>>>>>> or is something wrong with my dial string? >>>>>>> {origination_caller_id_number=12000}sofia/internal/ >>>>>>> 242424 at 192.168.1.62. >>>>>>> >>>>>>> In the logs, i cannot see that that my extension is being matched. >>>>>>> >>>>>>> Any idea, >>>>>>> >>>>>>> Max. >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/4cb9649c/attachment-0002.html From brian at freeswitch.org Mon Jun 22 09:06:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 12:06:25 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> Message-ID: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> originate {origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 &socket(192.168.50.67:10000 full) /b On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote: > Hi Mike, > > Unfortunately this doesn't seem to solve my problem. Here is my > extension again: > > > > > > > > I've copied it now under: > > /user/local/freeswitch/conf/dialplan/default > /user/local/freeswitch/conf/dialplan/public > > The different dial strings i tried: > > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 > & park()" > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67 > " > "{origination_caller_id_number=120003}sofia/internal/242424" > "{origination_caller_id_number=120003}sofia/internal/ > 242424%192.168.50.62" > > My goal: have the call captured by the above extension and > redirected to a server socket running at 192.168.50.67:10000. > > Any thought? > > Max. > > On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater > wrote: > I don't have my settings to try it right now. Still i have a > question. If it's the way you describe it, why wouldn't sofia/ > extenal/foo at bar solve the problem? I think i even copied the > extension both to the default directory. But i will confirm and let > you know. > > Max. > > > On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins > wrote: > Now I feel stupid because I didn't read your original post closely > enough. > > You've defined your "mysocket" extension in the "public" context but > when you do an origination with sofia/internal/foo at bar it will use > the "default" context. I think the quickest way to handle this is to > create a copy of your mysocket.xml file and put it in conf/dialplan/ > default/ and be done with it. > > -MC > > > On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater > wrote: > Mike, > > Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very > XML to me though. > > Max. > > > On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins > wrote: > Max, > that pastebin failed miserably as none of the xml shows up. can you > try again or use our pastebin.freeswitch.org site? > -MC > > > On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater > wrote: > Hi Mike, > > It's pasted here: http://pastebin.ca/1466521 > > Thanks, > Max. > > > > > On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins > wrote: > Can you turn on debugging (F8) and capture all the output after your > originate? Put it into a pastebin. (pastebin.freeswitch.org) > -MC > > On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater > wrote: > Any help our there? > > I'm still trying to get this piece working. Essentially what i wan > to do is, when a call comes in (from registered devices as well as > unregistered devices), notify the my server socket. Somehow it's not > working. The change i made compared to the standard Freeswitch > settings are the following: > > 1) Added following extension that in /usr/local/freeswitch/conf/ > dialplan/public/mysocket.xml: > > > > > > > > > > > 2) Changed file: /usr/local/freeswitch/conf/autoload_configs/ > event_socket.conf to: > > > > > > > > > > > > > I noticed that with this extension, all calls received from external > providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my > socket. But calls from registered devices and initiated using the > socket interface are not forwarded. Is there something that need to > be changed in the profiles? > > or is something wrong with my dial string? {origination_caller_id_number=12000}sofia/internal/242424 at 192.168.1.62 > . > > In the logs, i cannot see that that my extension is being matched. > > Any idea, > > Max. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b9ef667a/attachment-0002.html From max.bridgewater at gmail.com Mon Jun 22 10:08:51 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 22 Jun 2009 13:08:51 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> Message-ID: Hmm thamks. I tried it and it doesn't work out of the box. Here are my logs: http://pastebin.freeswitch.org/9454 Thanks, Max. On Mon, Jun 22, 2009 at 12:06 PM, Brian West wrote: > originate > {origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67<%7Borigination_caller_id_number=120003%7Dsofia/internal/242424 at 192.168.50.67>&socket( > 192.168.50.67:10000 full) > /b > > On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote: > > Hi Mike, > > Unfortunately this doesn't seem to solve my problem. Here is my extension > again: > > > > /> > > > > I've copied it now under: > > /user/local/freeswitch/conf/dialplan/default > /user/local/freeswitch/conf/dialplan/public > > The different dial strings i tried: > > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67& park()" > "{origination_caller_id_number=120003}sofia/internal/242424 at 192.168.50.67" > "{origination_caller_id_number=120003}sofia/internal/242424" > "{origination_caller_id_number=120003}sofia/internal/242424%192.168.50.62" > > My goal: have the call captured by the above extension and redirected to a > server socket running at 192.168.50.67:10000. > > Any thought? > > Max. > > On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> I don't have my settings to try it right now. Still i have a question. If >> it's the way you describe it, why wouldn't sofia/extenal/foo at bar solve >> the problem? I think i even copied the extension both to the default >> directory. But i will confirm and let you know. >> >> Max. >> >> >> On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins wrote: >> >>> Now I feel stupid because I didn't read your original post closely >>> enough. >>> >>> You've defined your "mysocket" extension in the "public" context but when >>> you do an origination with sofia/internal/foo at bar it will use the >>> "default" context. I think the quickest way to handle this is to create a >>> copy of your mysocket.xml file and put it in conf/dialplan/default/ and be >>> done with it. >>> >>> -MC >>> >>> >>> On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater < >>> max.bridgewater at gmail.com> wrote: >>> >>>> Mike, >>>> >>>> Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML >>>> to me though. >>>> >>>> Max. >>>> >>>> >>>> On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote: >>>> >>>>> Max, >>>>> that pastebin failed miserably as none of the xml shows up. can you try >>>>> again or use our pastebin.freeswitch.org site? >>>>> -MC >>>>> >>>>> >>>>> On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater < >>>>> max.bridgewater at gmail.com> wrote: >>>>> >>>>>> Hi Mike, >>>>>> >>>>>> It's pasted here: http://pastebin.ca/1466521 >>>>>> >>>>>> Thanks, >>>>>> Max. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins >>>>> > wrote: >>>>>> >>>>>>> Can you turn on debugging (F8) and capture all the output after your >>>>>>> originate? Put it into a pastebin. (pastebin.freeswitch.org) >>>>>>> -MC >>>>>>> >>>>>>> On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater < >>>>>>> max.bridgewater at gmail.com> wrote: >>>>>>> >>>>>>>> Any help our there? >>>>>>>> >>>>>>>> I'm still trying to get this piece working. Essentially what i wan >>>>>>>> to do is, when a call comes in (from registered devices as well as >>>>>>>> unregistered devices), notify the my server socket. Somehow it's not >>>>>>>> working. The change i made compared to the standard Freeswitch settings are >>>>>>>> the following: >>>>>>>> >>>>>>>> 1) Added following extension that in >>>>>>>> /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> > >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> 2) Changed file: >>>>>>>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> I noticed that with this extension, all calls received from external >>>>>>>> providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. >>>>>>>> But calls from registered devices and initiated using the socket interface >>>>>>>> are not forwarded. Is there something that need to be changed in the >>>>>>>> profiles? >>>>>>>> >>>>>>>> or is something wrong with my dial string? >>>>>>>> {origination_caller_id_number=12000}sofia/internal/ >>>>>>>> 242424 at 192.168.1.62. >>>>>>>> >>>>>>>> In the logs, i cannot see that that my extension is being matched. >>>>>>>> >>>>>>>> Any idea, >>>>>>>> >>>>>>>> Max. >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/6f165662/attachment-0002.html From brian at freeswitch.org Mon Jun 22 10:18:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 13:18:44 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906190843w6e292b36j2b77c34c1bfc043b@mail.gmail.com> <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> Message-ID: <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> what is 242424? If its a locally registered user you should be using a % instead of an @ /b On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: > Hmm thamks. I tried it and it doesn't work out of the box. Here are > my logs: http://pastebin.freeswitch.org/9454 > > Thanks, > Max. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/b9ebc105/attachment-0002.html From lon at kickasspixels.com Mon Jun 22 10:18:40 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 22 Jun 2009 10:18:40 -0700 Subject: [Freeswitch-users] Limit length of call with mod_limit? Message-ID: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> Hi there, Can mod_limit be used to restrict the length of a single call? I checked the wiki, dug into the code of mod_limit this weekend and couldn't find an answer. Lon From brian at freeswitch.org Mon Jun 22 10:22:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 13:22:11 -0400 Subject: [Freeswitch-users] Limit length of call with mod_limit? In-Reply-To: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> References: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Channel_Variables /b On Jun 22, 2009, at 1:18 PM, Lon Baker wrote: > Hi there, > > Can mod_limit be used to restrict the length of a single call? > > I checked the wiki, dug into the code of mod_limit this weekend and > couldn't find an answer. > > Lon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/deeed857/attachment-0002.html From raul at etellicom.com Mon Jun 22 10:44:18 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 22 Jun 2009 14:44:18 -0300 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: Message-ID: <1245692658.5598.19.camel@raul-laptop> I have recently updated the RPM spec for FreeSWITCH to use the latest SVN release, including some unspecified files for the newest mods (with nibblebil, unimrcp, etc, just like your build output is bitching about). Follow these steps to build it: 1 - Get the latest SVN release and make a tar-ball for it: $ cd /usr/src/redhat/SOURCES/ $ svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4trunk $ tar -cjvf freeswitch-1.0.4trunk.tar.bz2 freeswitch-1.0.4trunk/ 2 - Grab the libraries required to build FS: $ wget -q -O - http://www.etellicom.com/~raul/freeswitch_deps.txt | bash 3 - Download the RPM spec: $ cd /usr/src/redhat/SPECS/ $ wget http://www.etellicom.com/~raul/freeswitch.spec 4 - Build it :) $ rpmbuild -ba freeswitch.spec It works fine with a standard CentOS 5.3 installation. Regards, Raul On Mon, 2009-06-22 at 01:25 +0000, Joseph L. Casale wrote: > I attempted to build rpm's from the included spec file using a non-root user > build environment. The steps I used are as follows: > > 1. Check build deps @ http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS > 2. Pulled a copy of trunk in the SOURCES directory & tar/bzip2 it as expected by the spec: > svn co http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4 > tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/ > 3. Copy spec to SPECS directory: > 4. Pull in Source (Source0 doesn't exist yet, I make it above): > for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}' freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done > 5. Check spec in svn copy for deps: > yum install $(awk -v ORS=" " '/^BuildRequires:/ {print $2}' freeswitch.spec) > 6.Build rpm: > rpmbuild -ba freeswitch.spec > > After some time, near the end I see various issues like the following: > > making install mod_speex > installing mod_speex.so > quiet_libtool: install: warning: `/home/builder/rpmbuild/BUILD/freeswitch-1.0.4/libfreeswitch.la' > has not been installed in `/opt/freeswitch/lib' > > > It also seems to download everything it would normally again, then fails with > several errors like the following: > > RPM build errors: > File not found by glob: /var/tmp/freeswitch-1.0.4-1-root-builder/opt/freeswitch/mod/ozmod_wanpipe.so* > Installed (but unpackaged) file(s) found: > /opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml > /opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml > /opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml > /opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml > . > . > . > > After which no rpm's are built? Anyone know what tricks are still needed with > the spec from svn? > > Thanks! > jlc > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From max.bridgewater at gmail.com Mon Jun 22 11:01:29 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 22 Jun 2009 14:01:29 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> References: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> Message-ID: It's nothing. There is no extension like that. Shouldn't this nonetheless be caught by a regex such as the following? field="destination_number" expression="^242.*" The issue i have here is that it seems that the extensions aren't even processed. Usually, the log would show the list of processed extensions, each prefixed with the result "PASS", "FAIL". Max. On Mon, Jun 22, 2009 at 1:18 PM, Brian West wrote: > what is 242424? If its a locally registered user you should be using a % > instead of an @ > /b > > On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: > > Hmm thamks. I tried it and it doesn't work out of the box. Here are my > logs: http://pastebin.freeswitch.org/9454 > > Thanks, > Max. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/1345f913/attachment-0002.html From JCasale at activenetwerx.com Mon Jun 22 11:50:56 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 18:50:56 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> Message-ID: >Scroll up and look for the real error. All I see are these: *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libedit/src/.libs/libedit.a is not portable! ****** Integer sample type enabled ****** *** The gtk-config script installed by GTK could not be found *** If GTK was installed in PREFIX, make sure PREFIX/bin is in *** your path, or set the GTK_CONFIG environment variable to the *** full path to gtk-config. I am pulling down the newer libraries and updated spec now to try that. Thanks guys! jlc From brian at freeswitch.org Mon Jun 22 11:57:45 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 14:57:45 -0400 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: References: <4AEA82E9-41F3-4A98-AD90-77306F5D2584@jerris.com> <1E83BD5E-6855-497C-97F5-E7EAFF9947CE@freeswitch.org> Message-ID: <3CE93178-B052-4CCC-BE33-3DBA79661956@freeswitch.org> Ok, thats not the issue... look lower or post the full log. /b On Jun 22, 2009, at 2:50 PM, Joseph L. Casale wrote: > > All I see are these: > > *** Warning: Linking the shared library libfreeswitch.la against the > *** static library libs/libedit/src/.libs/libedit.a is not portable! > ****** Integer sample type enabled ****** > > *** The gtk-config script installed by GTK could not be found > *** If GTK was installed in PREFIX, make sure PREFIX/bin is in > *** your path, or set the GTK_CONFIG environment variable to the > *** full path to gtk-config. > > I am pulling down the newer libraries and updated spec now to try > that. > > Thanks guys! > jlc Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/26bca5c1/attachment-0002.html From timb0311 at hotmail.com Mon Jun 22 12:01:03 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 22 Jun 2009 15:01:03 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: Just wanted to follow-up again. Is this the proper or best way to configure this? See below... Tim From: timb0311 at hotmail.com To: freeswitch-users at lists.freeswitch.org Subject: (Found Fix) Transmit fax locally for test Date: Fri, 19 Jun 2009 23:06:10 -0400 Ok so after many attempts of trial and error I narrowed it down to acls. So when trying to orginate a call to the local FS extension it was getting blocked. Adding the following allow with my freeswitch IP to the domains list allowed the originate to take place. acl.conf.xml: So now this statement works for local fax testing: originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) Now my question is, is this the proper or best way to configure this? Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503 at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000 at x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users at lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > > haha, yeah i see it now... duh. pulled an all nighter, too many things > going on. must have overlooked it. > > > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > > the login and password is pastebin/freeswitch > > > > been trying to break myself into freeswitch on top of my original workload. > thanks for the help. > Bing? brings you maps, menus, and reviews organized in one place. Try it now. _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing? now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try bing_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/40b5079d/attachment-0002.html From brian at freeswitch.org Mon Jun 22 12:05:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 15:05:20 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67@freeswitch.org> what is 8000? is it local or is it a remote endpoint? /b On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > test.tif) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/f773841e/attachment-0002.html From drago at windstream.net Mon Jun 22 13:33:51 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 22 Jun 2009 16:33:51 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> Message-ID: <004e01c9f378$c1707e90$44517bb0$@net> Michael, It is definitely a problem in this build (13754M) (Windows) I found build from March and it does not brakes (still no joy, but not sure for now if the exchange side is configured properly). Again, I would submit bug report is you can point me to instructions how to do collect backtrace on Windows OS. Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Sunday, June 21, 2009 4:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 If this is still the case with current svn trunk, please get a backtrace and post it to http://jira.freeswitch.org On Jun 21, 2009, at 3:51 PM, "Drago Totev" wrote: After days of running around clueless, I have no other option but to ask the community for help one last time? Here is what happens: 1. FS sends INVITE 2. Exchange answers with ?302 Moved Temporarily? 3. FS bombs and closes Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: SIP to address: sip:4783874764 at 10.0.0.71 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 INVITE Contact: Contact Binding: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 236 P-Asserted-Identity: "MILLEDGEVL GA" Message Body Status-Line: SIP/2.0 100 Trying Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: SIP to address: sip:4783874764 at 10.0.0.71 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTENT-LENGTH: 0 Status-Line: SIP/2.0 302 Moved Temporarily Message Header FROM: "MILLEDGEVL GA";tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m TO: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 CSEQ: 116688296 INVITE CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF CONTACT: Contact Binding: CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF Max-Forwards: 68 From: "MILLEDGEVL GA" ;tag=Qr4j0XX18XD1m SIP Display info: "MILLEDGEVL GA" SIP from address: sip:4782512197 at 10.8.4.3 SIP tag: Qr4j0XX18XD1m To: ;tag=27df6afe0 SIP to address: sip:4783874764 at 10.0.0.71 SIP tag: 27df6afe0 Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 CSeq: 116688296 ACK Content-Length: 0 Here is the FS log beginning the the processing of the call: 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 [637f8a72-8034-254f-9937-b321c0d87414] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 entering state [received][100] 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 PCMU/8000 20 ms 160 samples 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf payload to 101 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_NEW 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59) State NEW 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_INIT 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 SOFIA INIT 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59) State INIT going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 SOFIA ROUTING 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 Standard ROUTING 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing MILLEDGEVL GA->4783874190 in context public Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->unloop] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->outside_call] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition [outside_call] Dialplan: sofia/external/4782512197 at 209.249.3.59 Action set(outside_call=true) Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->call_debug] continue=true Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [call_debug] ${call_debug}(true) =~ /^true$/ break=never Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->public_extensions] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public->Local_UM] continue=false Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on-false Dialplan: sofia/external/4782512197 at 209.249.3.59 Action bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59) State Change CS_ROUTING -> CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send signal sofia/external/4782512197 at 209.249.3.59 [BREAK] 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59) State ROUTING going to sleep 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59) Running State Change CS_EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59) State EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 SOFIA EXECUTE 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 Standard EXECUTE EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 SET [outside_call]=[true] EXECUTE sofia/external/4782512197 at 209.249.3.59 info() 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [4782512197] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MILLEDGEVL GA] Caller-Caller-ID-Number: [4782512197] Caller-Network-Addr: [209.249.3.59] Caller-Destination-Number: [4783874190] Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1245613758636018] Caller-Channel-Created-Time: [1245613758636018] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [209.249.3.59] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [4782512197] variable_sip_from_uri: [4782512197 at 209.249.3.59] variable_sip_from_host: [209.249.3.59] variable_sip_from_user_stripped: [4782512197] variable_sip_from_tag: [3454602382-411732] variable_sofia_profile_name: [external] variable_sip_cid_type: [pid] variable_sip_req_user: [gw+Broadvox] variable_sip_req_port: [5080] variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] variable_sip_req_host: [71.29.0.61] variable_sip_to_user: [4783874764] variable_sip_to_port: [5060] variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] variable_sip_to_host: [209.249.3.56] variable_sip_contact_user: [4782512197] variable_sip_contact_port: [5060] variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] variable_sip_contact_host: [209.249.3.59] variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] variable_sip_via_host: [209.249.3.59] variable_sip_via_port: [5060] variable_max_forwards: [69] variable_sip_call_info: [;method="NOTIFY;Event=telephone-event;Duration=1000"] variable_sip_gateway: [Broadvox] variable_switch_r_sdp: [v=0 o=NXT02 19785 8060 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 36292 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ] variable_remote_media_ip: [209.249.3.60] variable_remote_media_port: [36292] variable_read_codec: [PCMU] variable_read_rate: [8000] variable_write_codec: [PCMU] variable_write_rate: [8000] variable_endpoint_disposition: [RECEIVED] variable_outside_call: [true] variable_current_application: [info] EXECUTE sofia/external/4782512197 at 209.249.3.59 bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/4783874764) 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 variable string 0 = [absolute_codec_string=PCMA] 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/4783874764) State Change CS_NEW -> CS_INIT 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_INIT 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/4783874764 SOFIA INIT 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/4783874764) State Change CS_INIT -> CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/4783874764) State INIT going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/4783874764 SOFIA ROUTING 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/4783874764 [BREAK] 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/4783874764) State ROUTING going to sleep 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/4783874764) State CONSUME_MEDIA 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/internal/4783874764 entering state [calling][0] Will trade my first born for little help J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Drago Totev Sent: Sunday, June 14, 2009 7:11 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 Hello everyone, I am trying to set a test environment with FS and Exchange 2007. Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM and it does not seem to work. Thus far I use the default settings, one ATA registered and confirmed to work. According external source: ?MS Exchange does not understand 'From' fields with domain name in the address (example: field containing address).The ?From? field?s address to be in the format ??; otherwise, MS Exchange drops the call.? I don?t know if this is the only problem? However, I see exactly this behavior: ?PBX initiates call to MS Exchange by sending a SIP INVITE message to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved Temporarily) response to PBX asking to repeat INVITE to a different port (5065 for example). After PBX repeats the INVITE sending, the call is established.? After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without any record in the log. Can someone help with working configuration, please? Exchange 2007 UM role Version: 08.01.0359.002 Thanks in advance. Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/23f8e33f/attachment-0002.html From msc at freeswitch.org Mon Jun 22 13:44:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 22 Jun 2009 15:44:20 -0500 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906191110u1f55058euaa05bbaefdd72601@mail.gmail.com> <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> Message-ID: <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> On Mon, Jun 22, 2009 at 1:01 PM, Max Bridgewater wrote: > It's nothing. There is no extension like that. Shouldn't this nonetheless > be caught by a regex such as the following? > > field="destination_number" expression="^242.*" > > The issue i have here is that it seems that the extensions aren't even > processed. Usually, the log would show the list of processed extensions, > each prefixed with the result "PASS", "FAIL". > Max, if your originate line already has the sofia dialstring then there's really no reason to send the call through the dialplan - it already knows where to go. If you want to force the call through the dialplan then use loopback. However, you need some sort of endpoint for that to work. In your example you have this originate line: originate {origination_caller_id_number=120003}sofia/internal/ 242424 at 192.168.50.67 &socket(192.168.50.67:10000 full) Is 242424 at 192.168.50.67 a locally registered user? If so you could just do this: originate {origination_caller_id_number=120003} loopback/242424 &socket( 192.168.50.67:10000 full) This would run the A leg through the dialplan to look for destination number "242424" and then handle appropriately. If I understand your scenario I believe you are trying to get one leg of the call established and then the other leg handled by the event socket. What is the endpoint you want handled? A SIP phone that is registered locally? Or something else? In any case, you can CAN loop it through the dialplan but you aren't forced to do so. Assuming 1000 is locally registered: originate {origination_caller_id_number=120003} sofia/internal/1000%192.168.50.67 &socket(192.168.50.67:10000 full) originate {origination_caller_id_number=120003} user/1000 &socket( 192.168.50.67:10000 full) originate {origination_caller_id_number=120003} loopback/1000 &socket( 192.168.50.67:10000 full) NOTE: the first two do not use the dialplan but the third example does. This means you MUST handle destination_number="1000" in your dialplan (which the default config does). Hope this helps. -MC > Max. > > On Mon, Jun 22, 2009 at 1:18 PM, Brian West wrote: > >> what is 242424? If its a locally registered user you should be using a % >> instead of an @ >> /b >> >> On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: >> >> Hmm thamks. I tried it and it doesn't work out of the box. Here are my >> logs: http://pastebin.freeswitch.org/9454 >> >> Thanks, >> Max. >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/2f970056/attachment-0002.html From larclap at yahoo.com Mon Jun 22 13:49:16 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 22 Jun 2009 13:49:16 -0700 Subject: [Freeswitch-users] Polycom configuration problems? Message-ID: <003f01c9f37a$e8b70d00$ba252700$@com> I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/a7c44c1e/attachment-0002.html From chris at cloudtel.com Mon Jun 22 14:57:23 2009 From: chris at cloudtel.com (Chris Burns) Date: Mon, 22 Jun 2009 16:57:23 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <003f01c9f37a$e8b70d00$ba252700$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> Message-ID: Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or .digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 > lines on the phone. The first two are registered with a SwitchVox, the last > with Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/cb52cac0/attachment-0002.html From timb0311 at hotmail.com Mon Jun 22 15:37:47 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 22 Jun 2009 18:37:47 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: 8000 is a local extension defined in the default dialplan. Tim > ------------------------------ > > Message: 2 > Date: Mon, 22 Jun 2009 15:05:20 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > what is 8000? is it local or is it a remote endpoint? > > /b > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > test.tif) > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > _________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/5ab527bb/attachment-0002.html From jim at evolutiontel.net Mon Jun 22 15:53:22 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 23 Jun 2009 08:53:22 +1000 Subject: [Freeswitch-users] Limit length of call with mod_limit? In-Reply-To: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> References: <5d3e0dc60906221018y5a707613y228076852f4925b3@mail.gmail.com> Message-ID: This line in your diaplan will set a timer to hangup the calls x secs after answer. On Tue, Jun 23, 2009 at 3:18 AM, Lon Baker wrote: > Hi there, > > Can mod_limit be used to restrict the length of a single call? > > I checked the wiki, dug into the code of mod_limit this weekend and > couldn't find an answer. > > Lon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/d0c15ac1/attachment-0002.html From JCasale at activenetwerx.com Mon Jun 22 15:55:17 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 22 Jun 2009 22:55:17 +0000 Subject: [Freeswitch-users] rpm build issues on CentOS 5 with trunk In-Reply-To: <1245692658.5598.19.camel@raul-laptop> References: <1245692658.5598.19.camel@raul-laptop> Message-ID: >I have recently updated the RPM spec for FreeSWITCH to use the latest >SVN release, including some unspecified files for the newest mods (with >nibblebil, unimrcp, etc, just like your build output is bitching about). >Follow these steps to build it: Raul, Appreciate this, it worked. I am re-running it as the first time I was only logging stdout and it appears there are some errors in my build worth exploring. Nothing prevented the build, so I created a local repo with all the rpm's and executed a `yum install freeswitch` and I see that it never pulled in anything else. Where abouts in the docs could I find info on knowing what I need for an initial install to test? Thanks for the help! jlc From jingwei.yang at gmail.com Mon Jun 22 19:33:09 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 23 Jun 2009 10:33:09 +0800 Subject: [Freeswitch-users] How to originate gtalk calls Message-ID: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch> *originate dingaling/gmail.com/userAAA at gmail.com &echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* Please kindly let me know what the correct originate string is. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/a8b8d86b/attachment-0002.html From brian at freeswitch.org Mon Jun 22 20:10:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 22:10:42 -0500 Subject: [Freeswitch-users] How to originate gtalk calls In-Reply-To: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Message-ID: <5AB2BF21-119D-478E-B582-229E1AD91D1B@freeswitch.org> Might need to compile and load mod_dingaling first. /b On Jun 22, 2009, at 9:33 PM, Jingwei Yang wrote: > Hi Guys, > > I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ > . > > But i'm not sure how to originate calls to different gtalk users > dynamically. I've tried this: > > freeswitch> originate dingaling/gmail.com/userAAA at gmail.com &echo > > but got CHAN_NOT_IMPLEMENTED error. > > 2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot > create outgoing channel of type [dingaling] cause: > [CHAN_NOT_IMPLEMENTED] > > Please kindly let me know what the correct originate string is. > Thanks! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/4a47413e/attachment-0002.html From mashudiflexi at telkom.co.id Mon Jun 22 20:28:07 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Tue, 23 Jun 2009 10:28:07 +0700 Subject: [Freeswitch-users] video playback on FS In-Reply-To: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Message-ID: <4A404BC7.2060403@telkom.co.id> Dear All, on the default.xml in dialplan directory of FS, content video extension dialplan with file extension fsv, 562 563 564 565 566 567 568 569 570 571 572 573 574 what is the fsv video format from? as we know flv for flash video, how to convert from mp4 or avi to fsv file extension? thank you in advanced, best regard, mashudi ==================================== Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya: - hubungi 147 - http://www.telkomflexi.com - ketik INFO, sms ke 345. From jmesquita at gmail.com Mon Jun 22 20:14:43 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 23 Jun 2009 00:14:43 -0300 Subject: [Freeswitch-users] How to originate gtalk calls In-Reply-To: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> Message-ID: <5a8712120906222014s43c220b0x29ff0c3e6e790fb0@mail.gmail.com> try load mod_dingaling. If that does not work, get to the source dir, edit modules.conf, uncomment mod_dingaling, make && make install Dont forget to load the mod once FS is up again.. jmesquita On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang wrote: > Hi Guys, > > I've configured a gtalk client based on the steps in this url: > http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. > > But i'm not sure how to originate calls to different gtalk users > dynamically. I've tried this: > > freeswitch> *originate dingaling/gmail.com/userAAA at gmail.com &echo* > > but got CHAN_NOT_IMPLEMENTED error. > > *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create > outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* > > Please kindly let me know what the correct originate string is. Thanks! > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/3fc8b1fc/attachment-0002.html From jingwei.yang at gmail.com Mon Jun 22 20:25:08 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 23 Jun 2009 11:25:08 +0800 Subject: [Freeswitch-users] How to originate gtalk calls In-Reply-To: <5a8712120906222014s43c220b0x29ff0c3e6e790fb0@mail.gmail.com> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> <5a8712120906222014s43c220b0x29ff0c3e6e790fb0@mail.gmail.com> Message-ID: <13529f9d0906222025v781dbc80qd3ca9d1179730282@mail.gmail.com> Hi Brian and Jo?o, you're right, I forgot to load mod_dingaling. Thanks for the help. 2009/6/23 Jo?o Mesquita > try load mod_dingaling. > > If that does not work, get to the source dir, edit modules.conf, uncomment > mod_dingaling, make && make install > > Dont forget to load the mod once FS is up again.. > > jmesquita > > On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang wrote: > >> Hi Guys, >> >> I've configured a gtalk client based on the steps in this url: >> http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. >> >> But i'm not sure how to originate calls to different gtalk users >> dynamically. I've tried this: >> >> freeswitch> *originate dingaling/gmail.com/userAAA at gmail.com &echo* >> >> but got CHAN_NOT_IMPLEMENTED error. >> >> *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot >> create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED] >> * >> >> Please kindly let me know what the correct originate string is. Thanks! >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/7e47b81b/attachment-0002.html From brian at freeswitch.org Mon Jun 22 20:26:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 22 Jun 2009 22:26:18 -0500 Subject: [Freeswitch-users] video playback on FS In-Reply-To: <4A404BC7.2060403@telkom.co.id> References: <13529f9d0906221933t2bde2b52maadf6348fcd0a272@mail.gmail.com> <4A404BC7.2060403@telkom.co.id> Message-ID: <7B3175DB-042E-45BB-BD87-7DC5573B8622@freeswitch.org> On Jun 22, 2009, at 10:28 PM, mashudi wrote: > Dear All, > on the default.xml in dialplan directory of FS, content video > extension > dialplan with file extension fsv, > > 562 > 563 expression="^9993$"> > 564 > 565 data="/tmp/testrecord.fsv"/> > 566 > 567 > 568 > 569 > 570 expression="^9994$"> > 571 > 572 > 573 > 574 > > what is the fsv video format from? as we know flv for flash video, > how to convert from mp4 or avi to fsv file extension? We just save the raw RTP and stream it back out... btw don't hijack threads please. > thank you in advanced, > > best regard, > > mashudi Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090622/36b08c36/attachment-0002.html From darklion11 at yahoo.com Mon Jun 22 20:38:24 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 20:38:24 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED In-Reply-To: <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> References: <24143545.post@talk.nabble.com> <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> Message-ID: <24158819.post@talk.nabble.com> Nope. I just want to call a mobile number with no register number. Brian West-3 wrote: > > I'm going to guess you're calling a registered user? If so replace > the @ with % > > /b > > On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: > >> >> Hi, >> >> API CALL [originate sofia/external/1001 at 116.50.456.212] >> -ERR SERVICE_NOT_IMPLEMENTED >> >> I receiving this error i dont know y? Can u help mo on this? >> >> I dialing a mobile number on this sometimes it works... Sometimes it >> destroys the call [CALL_DESTROY] >> >> >> Thanks > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24158819.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Mon Jun 22 21:02:49 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 21:02:49 -0700 (PDT) Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 Message-ID: <24158823.post@talk.nabble.com> When I calling an outbound extension it appears: name is FreeSWITCH and number is 000000000 How can i change it depends on the user who is calling? Sample 1001->64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From q.edward at gmail.com Mon Jun 22 21:26:32 2009 From: q.edward at gmail.com (Edward Q.) Date: Tue, 23 Jun 2009 00:26:32 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <24158823.post@talk.nabble.com> References: <24158823.post@talk.nabble.com> Message-ID: <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: > > When I calling an outbound extension it appears: > > name is FreeSWITCH and number is 000000000 > > How can i change it depends on the user who is calling? > > Sample 1001->64521223 > > I just want the name 1001 to appear not FreeSWITCH same as the number > > Thanks > > > > -- > View this message in context: > http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/176d1862/attachment-0002.html From otrcomm at isp-systems.net Mon Jun 22 20:31:54 2009 From: otrcomm at isp-systems.net (murrah boswell) Date: Mon, 22 Jun 2009 20:31:54 -0700 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment Message-ID: <4A404CAA.5080809@isp-systems.net> Hello All, I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance on how to setup a testbed in a thin client environment. I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and utilize fl_teachertool 0.07 to monitor the connected terminal clients (TCs). If you are not familiar with fl_teachertool, it allows a teacher to view thumbnail images of each TC logged in to the server. The teacher can click on any thumbnail and enlarge the view, monitor all applications running on a given TC, and take control of the keyboard and mouse of the TC. These are just a few of the capabilities of fl_teachertool. What I want to do is allow the teacher to establish voice communication using headsets and microphones with any one of the TCs by making a "phone" call via ethernet based upon ip of the TC through freeswitch using a softphone. Does this sound like something that is possible using freeswitch? If so, could someone please give me very basic instructions on how to setup this proof of concept? If I can just get a "teacher" stationed at my server talking to one "student" at a TC, I believe I can go from there. Currently I have a voiper softphone that functions, I believe, under gnome, but I have no idea how to configure the voiper to initiate calls through freeswitch or how to configure freeswitch to route the call to one of my TCs. I also need to keep this system fully self contained. That is, I can not have a requirement to use an outside sip service provider. Also, I would use any other linux sip softphones known to work with freeswitch that people feel would work better than a voiper. voiper seems to be more windows and mac based. I would really like to use an ekiga since they seem to be more linux based, but I do not believe that they have been thoroughly tested with freeswitch. Any help would be greatly appreciated! Regards, Murrah Boswell From vince.freeswitch at hightek.org Mon Jun 22 21:08:53 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Mon, 22 Jun 2009 23:08:53 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD Message-ID: <20090623040853.GA84157@quark.hightek.org> Hi. I have been searching for an alternative PBX to asterisk (which has not been all that stable) to run on Dragonfly BSD. I spent a fair amount of time a couple months ago trying to compile freeswitch without success. I have since tried Yate, but it consumes 85-95% of the cpu when idle (not processing any calls). I am considering revisiting freeswitch. I have included below my notes of all the various problems I encountered before, which ones I resolved and how, up to the point where I left off. I am in hopes of getting feedback on whether any of the issues have been fixed or are planned to be fixed and/or suggestions on getting it working. Also, perhaps these notes on my experiences will be helpful for the developers to improve freeswitch. Keep in mind that Dragonfly is a branch from Freebsd and stays fairly compatible. I am able to compile most software, that is ported to FreeBSD, with few problems. Here are my notes ================= To compile on dragonfly BSD 1.10.1-RELEASE ========================================== I had to add -D__FreeBSD__ to CPPFLAGS ln sh to bash or zsh because I got "unexpected operator" errors from "test" during configure with the bsd shell. ln make to gmake Their scripts were calling make even though I ran the build using gmake. Must have apr-0.9.16.2.0.61 and apr-util-0.9.16.2.0.61 installed apr-0.xxx has headers, which freeswitch is including, that apr-1.xxx does not have. I sym-linked the apr-util libs from apr-util-1.2.8nb1 because apr-util-0.9.16.2.0.61 was not available as a binary package. The freeswitch-1.0.3.tar.gz release did not have bootstrap.sh, which is required for building from the svn repository. Somebody on the #freeswitch IRC suggested I get it from the subversion repository and run the bootstrap script. bootstrap.sh is not in the release. ========================================== Tue Mar 31 00:12:37 CDT 2009 I posted on the freeswitch mailing list asking about the compilation errors Got no responses. This first set of apr_... warnings turned out to clearly be from not having the correct apr-util package installed, I should have gotten a response on the list about it, considering it is a clear dependency that they do not directly specify on the web site or the source docs. apr-util is a dependency of subversion. They list SVN as a dependency of freeswitch, which is the utility in the subversion package, not the package name. svn should not be a dependency to build from a release archive that is not retrieved from the svn repository. As it turns out, it had to be apr-util version 0.9.15. See notes below. ========================================== Tue Mar 31 22:22:58 CDT 2009 I tried the freeswitch-snapshot.tar.gz from the freeswitch site, which is a 03/30/2009 snapshot from the svn trunk. *concern* It was nearly twice as big as the release for some reason. 27016871 Mar 28 13:08 freeswitch-1.0.3.tar.gz 52854882 Mar 31 18:25 freeswitch-snapshot.tar.gz Running bootstrap.sh produced a bunch of these errors from automake: Use of uninitialized value in exists at /usr/pkg/bin/automake line 4823, line 1. Use of uninitialized value in concatenation (.) or string at /usr/pkg/bin/automake line 4823, line 1. automake: #################### automake: ## Internal Error ## automake: #################### automake: unrequested trace `' automake: Please contact . at /usr/pkg/share/automake-1.10/Automake/Channels.pm line 570 Automake::Channels::msg('automake', '', 'unrequested trace `\'') called at /usr/pkg/share/automake-1.10/Automake/ChannelDefs.pm line 191 Automake::ChannelDefs::prog_error('unrequested trace `\'') called at /usr/pkg/bin/automake line 4823 Automake::scan_autoconf_traces('configure.ac') called at /usr/pkg/bin/automake line 5046 Automake::scan_autoconf_files() called at /usr/pkg/bin/automake line 781 ========================================== Thu Apr 2 20:51:44 CDT 2009 Going back to the 1.0.3 release. I was getting a a bunch of apr_... warnings like Compiling src/switch_apr.c ... src/switch_apr.c: In function `switch_thread_self': src/switch_apr.c:74: warning: implicit declaration of function `apr_os_thread_current' src/switch_apr.c:74: warning: return makes pointer from integer without a cast ... Then gmake[3]: *** [libfreeswitch_la-switch_apr.lo] Error 1 I finally got past that by installing apr-0.9.16.2.0.61 and apr-util-0.9.15. I compiled apr-util-0.9.15 myself because it was not available as a binary package. Source files are including headers from apr-util-0.9.15 that do not exist in the binary package, apr-util-1.2.8nb1.tgz. I also got past the "Cannot guess build type" error during configure by adding "--build=i386" instead of having to use the uname wrapper to fake FreeBSD. ========================================== New error: (This one is not the fault of freeswitch) Compiling src/switch_core.c ... src/switch_core.c: In function `switch_core_setrlimits': src/switch_core.c:795: error: `RLIMIT_AS' undeclared (first use in this function) Turns out RLIMIT_AS is defined in /usr/include/sys/resource.h on some other systems including FreeBSD but not Dragonfly. It is used in the source like this #if !defined(__OpenBSD__) && !defined(__NetBSD__) setrlimit(RLIMIT_AS, &rlp); It is defined in resource.h in FreeBSD like this #define RLIMIT_AS RLIMIT_VMEM /* standard name for RLIMIT_VMEM */ I created a patch for src/switch_core.c which fixed this error. Since then, I discussed this on the Dragonfly BSD mailing list and it is apparently going to be fixed on a future release of Dragonfly now that they are aware of it. ========================================== New error: Tons of warnings like warning: return makes pointer from integer without a cast (this needs cleaned up) Then error'd with gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. I cd'd to libs/sofia-sip/libsofia-sip-ua/su and ran make there. It completed successfully in spite of the same warnings. Then ran make again form the top. Got further until ========================================== New error: making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ports/freeswitch-1.0.3/work/freeswitch-1.0.3/libs/js/nsprpub/dist/include/nspr/.: File exists Why would it be trying to make a sym-link of '.' ?? ========================================== Giving up for now. From harmeet at litatel.com Mon Jun 22 21:46:22 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Tue, 23 Jun 2009 00:46:22 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Message-ID: In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > Edmar > > Eso esta en freeswitch/conf/vars.xml en ese archivo. > If i am not mistaken and anyone welcome to correct me i just told Edmar > this is set in freeswitch/conf/vars.xml ... file > Ed > > > On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: > >> >> When I calling an outbound extension it appears: >> >> name is FreeSWITCH and number is 000000000 >> >> How can i change it depends on the user who is calling? >> >> Sample 1001->64521223 >> >> I just want the name 1001 to appear not FreeSWITCH same as the number >> >> Thanks >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/bf321631/attachment-0002.html From harmeet at litatel.com Mon Jun 22 21:47:46 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Tue, 23 Jun 2009 00:47:46 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Message-ID: Oops, that was too quick! In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - Thanks On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh wrote: > In my case the 1001 resides in - > /usr/local/freeswitch/conf/directory/default/1001.xml > > And you set the Caller Name and ID by adding - > > > > > > > On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > >> Edmar >> >> Eso esta en freeswitch/conf/vars.xml en ese archivo. >> If i am not mistaken and anyone welcome to correct me i just told Edmar >> this is set in freeswitch/conf/vars.xml ... file >> Ed >> >> >> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: >> >>> >>> When I calling an outbound extension it appears: >>> >>> name is FreeSWITCH and number is 000000000 >>> >>> How can i change it depends on the user who is calling? >>> >>> Sample 1001->64521223 >>> >>> I just want the name 1001 to appear not FreeSWITCH same as the number >>> >>> Thanks >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/8a5f05f7/attachment-0002.html From q.edward at gmail.com Mon Jun 22 23:06:42 2009 From: q.edward at gmail.com (Edward Q.) Date: Tue, 23 Jun 2009 02:06:42 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> Message-ID: <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> Sorry Edmar I missundertood you .. I thought you wanted to change the number showing once you were going out not the 1001.xml file. In this case Harmeet is right. There you have those values to to make the changes. My bad. Ed On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh wrote: > In my case the 1001 resides in - > /usr/local/freeswitch/conf/directory/default/1001.xml > > And you set the Caller Name and ID by adding - > > > > > > > On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > >> Edmar >> >> Eso esta en freeswitch/conf/vars.xml en ese archivo. >> If i am not mistaken and anyone welcome to correct me i just told Edmar >> this is set in freeswitch/conf/vars.xml ... file >> Ed >> >> >> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz wrote: >> >>> >>> When I calling an outbound extension it appears: >>> >>> name is FreeSWITCH and number is 000000000 >>> >>> How can i change it depends on the user who is calling? >>> >>> Sample 1001->64521223 >>> >>> I just want the name 1001 to appear not FreeSWITCH same as the number >>> >>> Thanks >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/b0b39470/attachment-0002.html From darklion11 at yahoo.com Mon Jun 22 23:39:39 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 22 Jun 2009 23:39:39 -0700 (PDT) Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> Message-ID: <24160712.post@talk.nabble.com> Actually the extension_caller_id=Extension 1001 and extension_caller_number=1001 is set as Harmeet says but the same issue FreeSwitch the caller name and the number is 0000000 i just want 1001 the caller number and the id Edmar Edward Q. wrote: > > Sorry Edmar > > I missundertood you .. I thought you wanted to change the number showing > once you were going out not the 1001.xml file. > In this case Harmeet is right. There you have those values to to make the > changes. > My bad. > Ed > > On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh > wrote: > >> In my case the 1001 resides in - >> /usr/local/freeswitch/conf/directory/default/1001.xml >> >> And you set the Caller Name and ID by adding - >> >> >> >> >> >> >> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: >> >>> Edmar >>> >>> Eso esta en freeswitch/conf/vars.xml en ese archivo. >>> If i am not mistaken and anyone welcome to correct me i just told Edmar >>> this is set in freeswitch/conf/vars.xml ... file >>> Ed >>> >>> >>> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz >>> wrote: >>> >>>> >>>> When I calling an outbound extension it appears: >>>> >>>> name is FreeSWITCH and number is 000000000 >>>> >>>> How can i change it depends on the user who is calling? >>>> >>>> Sample 1001->64521223 >>>> >>>> I just want the name 1001 to appear not FreeSWITCH same as the number >>>> >>>> Thanks >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24160712.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Tue Jun 23 04:03:54 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 23 Jun 2009 19:03:54 +0800 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <24160712.post@talk.nabble.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> <24160712.post@talk.nabble.com> Message-ID: <1AEB5722-06B2-4B3F-858B-164E10008C5B@gmail.com> depending how do you make out going call. On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote: > > Actually the extension_caller_id=Extension 1001 and > extension_caller_number=1001 is set as Harmeet says but the same issue > FreeSwitch the caller name and the number is 0000000 i just want > 1001 the > caller number and the id > > Edmar > > Edward Q. wrote: >> >> Sorry Edmar >> >> I missundertood you .. I thought you wanted to change the number >> showing >> once you were going out not the 1001.xml file. >> In this case Harmeet is right. There you have those values to to >> make the >> changes. >> My bad. >> Ed >> >> On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh >> wrote: >> >>> In my case the 1001 resides in - >>> /usr/local/freeswitch/conf/directory/default/1001.xml >>> >>> And you set the Caller Name and ID by adding - >>> >>> >>> >>> >>> >>> >>> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. >>> wrote: >>> >>>> Edmar >>>> >>>> Eso esta en freeswitch/conf/vars.xml en ese archivo. >>>> If i am not mistaken and anyone welcome to correct me i just told >>>> Edmar >>>> this is set in freeswitch/conf/vars.xml ... file >>>> Ed >>>> >>>> >>>> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> When I calling an outbound extension it appears: >>>>> >>>>> name is FreeSWITCH and number is 000000000 >>>>> >>>>> How can i change it depends on the user who is calling? >>>>> >>>>> Sample 1001->64521223 >>>>> >>>>> I just want the name 1001 to appear not FreeSWITCH same as the >>>>> number >>>>> >>>>> Thanks >>>>> >>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24160712.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Tue Jun 23 05:51:30 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 23 Jun 2009 22:51:30 +1000 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? Message-ID: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks! From larclap at yahoo.com Tue Jun 23 06:26:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 23 Jun 2009 06:26:38 -0700 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> Message-ID: <005401c9f406$3ced3710$b6c7a530$@com> I'm sorry Chris, but I don't know where the look for the "global sip.cfg and mac/phone specific cfg" settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or .digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/da144c31/attachment-0002.html From harmeet at litatel.com Tue Jun 23 06:27:34 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Tue, 23 Jun 2009 09:27:34 -0400 Subject: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000 In-Reply-To: <24160712.post@talk.nabble.com> References: <24158823.post@talk.nabble.com> <89313a90906222126m26c6e861t74a5e8329eb62b5b@mail.gmail.com> <89313a90906222306i685625f0l208ac6dc241b1061@mail.gmail.com> <24160712.post@talk.nabble.com> Message-ID: Check your dialplan where you call "bridge" to gateway to make outgoing calls. Stick in the following lines before the bridge call - On Tue, Jun 23, 2009 at 2:39 AM, Edmar Cruz wrote: > > Actually the extension_caller_id=Extension 1001 and > extension_caller_number=1001 is set as Harmeet says but the same issue > FreeSwitch the caller name and the number is 0000000 i just want 1001 the > caller number and the id > > Edmar > > Edward Q. wrote: > > > > Sorry Edmar > > > > I missundertood you .. I thought you wanted to change the number showing > > once you were going out not the 1001.xml file. > > In this case Harmeet is right. There you have those values to to make the > > changes. > > My bad. > > Ed > > > > On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh > > wrote: > > > >> In my case the 1001 resides in - > >> /usr/local/freeswitch/conf/directory/default/1001.xml > >> > >> And you set the Caller Name and ID by adding - > >> > >> > >> > >> > >> > >> > >> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. wrote: > >> > >>> Edmar > >>> > >>> Eso esta en freeswitch/conf/vars.xml en ese archivo. > >>> If i am not mistaken and anyone welcome to correct me i just told Edmar > >>> this is set in freeswitch/conf/vars.xml ... file > >>> Ed > >>> > >>> > >>> On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz > >>> wrote: > >>> > >>>> > >>>> When I calling an outbound extension it appears: > >>>> > >>>> name is FreeSWITCH and number is 000000000 > >>>> > >>>> How can i change it depends on the user who is calling? > >>>> > >>>> Sample 1001->64521223 > >>>> > >>>> I just want the name 1001 to appear not FreeSWITCH same as the number > >>>> > >>>> Thanks > >>>> > >>>> > >>>> > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24158823.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D000000000-tp24158823p24160712.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/481a91d0/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 23 06:29:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 08:29:13 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090623040853.GA84157@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> Message-ID: <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> You are way off base in a few places, let me see if I can clarify a bit. Here are at least 2 pointers: 1) The release tarballs do not come with bootstrap because they already are bootstrapped. 2) FreeSWITCH does not depend on system libs so all the stuff about apr is barking up the wrong tree. we build our own apr and apr-utils I suggest you try latest svn trunk of FS and follow the BSD build guidelines on the WIKI since you say it's closely compatible. On Mon, Jun 22, 2009 at 11:08 PM, Vincent Stemen < vince.freeswitch at hightek.org> wrote: > Hi. > > I have been searching for an alternative PBX to asterisk (which has not > been all that stable) to run on Dragonfly BSD. I spent a fair amount of > time a couple months ago trying to compile freeswitch without success. > > I have since tried Yate, but it consumes 85-95% of the cpu when idle > (not processing any calls). > > I am considering revisiting freeswitch. I have included below my notes > of all the various problems I encountered before, which ones I resolved > and how, up to the point where I left off. I am in hopes of getting > feedback on whether any of the issues have been fixed or are planned to > be fixed and/or suggestions on getting it working. Also, perhaps these > notes on my experiences will be helpful for the developers to improve > freeswitch. > > Keep in mind that Dragonfly is a branch from Freebsd and stays fairly > compatible. I am able to compile most software, that is ported to > FreeBSD, with few problems. > > Here are my notes > ================= > > To compile on dragonfly BSD 1.10.1-RELEASE > ========================================== > I had to add -D__FreeBSD__ to CPPFLAGS > ln sh to bash or zsh > because I got "unexpected operator" errors from "test" during configure > with the bsd shell. > ln make to gmake > Their scripts were calling make even though I ran the build > using gmake. > Must have apr-0.9.16.2.0.61 and apr-util-0.9.16.2.0.61 installed > apr-0.xxx has headers, which freeswitch is including, that apr-1.xxx > does not have. > I sym-linked the apr-util libs from apr-util-1.2.8nb1 because > apr-util-0.9.16.2.0.61 was not available as a binary package. > > The freeswitch-1.0.3.tar.gz release did not have bootstrap.sh, which is > required for building from the svn repository. > > Somebody on the #freeswitch IRC suggested I get it from the subversion > repository and run the bootstrap script. bootstrap.sh is not in the > release. > > ========================================== > Tue Mar 31 00:12:37 CDT 2009 > > I posted on the freeswitch mailing list asking about the compilation errors > Got no responses. > > This first set of apr_... warnings turned out to clearly be from not having > the > correct apr-util package installed, I should have gotten a response on the > list > about it, considering it is a clear dependency that they do not directly > specify on the web site or the source docs. apr-util is a dependency of > subversion. They list SVN as a dependency of freeswitch, which is the > utility > in the subversion package, not the package name. svn should not be > a dependency to build from a release archive that is not retrieved from the > svn > repository. > > As it turns out, it had to be apr-util version 0.9.15. See notes below. > ========================================== > > Tue Mar 31 22:22:58 CDT 2009 > > I tried the freeswitch-snapshot.tar.gz from the freeswitch site, which is > a 03/30/2009 snapshot from the svn trunk. > > *concern* > It was nearly twice as big as the release for some reason. > 27016871 Mar 28 13:08 freeswitch-1.0.3.tar.gz > 52854882 Mar 31 18:25 freeswitch-snapshot.tar.gz > > Running bootstrap.sh produced a bunch of these errors from automake: > > Use of uninitialized value in exists at /usr/pkg/bin/automake line 4823, > line 1. > Use of uninitialized value in concatenation (.) or string at > /usr/pkg/bin/automake line 4823, line 1. > automake: #################### > automake: ## Internal Error ## > automake: #################### > automake: unrequested trace `' > automake: Please contact . > at /usr/pkg/share/automake-1.10/Automake/Channels.pm line 570 > Automake::Channels::msg('automake', '', 'unrequested trace `\'') > called at /usr/pkg/share/automake-1.10/Automake/ChannelDefs.pm line 191 > Automake::ChannelDefs::prog_error('unrequested trace `\'') called at > /usr/pkg/bin/automake line 4823 > Automake::scan_autoconf_traces('configure.ac') called at > /usr/pkg/bin/automake line 5046 > Automake::scan_autoconf_files() called at /usr/pkg/bin/automake line > 781 > > > ========================================== > Thu Apr 2 20:51:44 CDT 2009 > > Going back to the 1.0.3 release. > > I was getting a a bunch of apr_... warnings like > > Compiling src/switch_apr.c ... > src/switch_apr.c: In function `switch_thread_self': > src/switch_apr.c:74: warning: implicit declaration of function > `apr_os_thread_current' > src/switch_apr.c:74: warning: return makes pointer from integer without > a cast > ... > Then > gmake[3]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > > I finally got past that by installing apr-0.9.16.2.0.61 and > apr-util-0.9.15. > I compiled apr-util-0.9.15 myself because it was not available as a binary > package. > Source files are including headers from apr-util-0.9.15 that do not exist > in the > binary package, apr-util-1.2.8nb1.tgz. > > I also got past the "Cannot guess build type" error during configure by > adding > "--build=i386" instead of having to use the uname wrapper to fake FreeBSD. > > ========================================== > New error: (This one is not the fault of freeswitch) > > Compiling src/switch_core.c ... > src/switch_core.c: In function `switch_core_setrlimits': > src/switch_core.c:795: error: `RLIMIT_AS' undeclared (first use in this > function) > > Turns out RLIMIT_AS is defined in /usr/include/sys/resource.h on some other > systems including FreeBSD but not Dragonfly. > > It is used in the source like this > #if !defined(__OpenBSD__) && !defined(__NetBSD__) > setrlimit(RLIMIT_AS, &rlp); > > It is defined in resource.h in FreeBSD like this > #define RLIMIT_AS RLIMIT_VMEM /* standard name for RLIMIT_VMEM > */ > > I created a patch for src/switch_core.c which fixed this error. > > Since then, I discussed this on the Dragonfly BSD mailing list and it is > apparently going to be fixed on a future release of Dragonfly now that > they are aware of it. > > ========================================== > New error: > > Tons of warnings like > warning: return makes pointer from integer without a cast > (this needs cleaned up) > > Then error'd with > gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by ` > libsofia-sip-ua.la'. Stop. > > I cd'd to libs/sofia-sip/libsofia-sip-ua/su > and ran make there. It completed successfully in spite of the same > warnings. > > Then ran make again form the top. > > Got further until > > ========================================== > New error: > > making all mod_spidermonkey > cd config; gmake -j1 export > cd pr; gmake -j1 export > cd include; gmake export > cd md; gmake export > ../../../config/./nsinstall: cannot make symbolic link > /u1/falcon/ports/freeswitch-1.0.3/work/freeswitch-1.0.3/libs/js/nsprpub/dist/include/nspr/.: > File exists > > Why would it be trying to make a sym-link of '.' ?? > > ========================================== > > Giving up for now. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/62794fec/attachment-0002.html From Claudio.Cavalera at italtel.it Tue Jun 23 06:33:02 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Tue, 23 Jun 2009 15:33:02 +0200 Subject: [Freeswitch-users] Variable manipulation in the dialplan Message-ID: Hello, I once found in the wiki a page explaining how to "substring" a channel variable, something like <@[intra]lanman> 12345 would be 345 if you do ${var:2} I can't find that page on the wiki anymore, any hint on were it could be? :-) Also do you think it could be useful to extend this functionality with a sort of Java indexOf() to extract a specific substring from a variable (but without knowing its size like in the example above)? Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From anthony.minessale at gmail.com Tue Jun 23 06:44:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 08:44:43 -0500 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Message-ID: <191c3a030906230644i154a6198j2cf1027e688297d3@mail.gmail.com> try adding this to your jingle in client.xml then edit acl.conf.xml and add this list this tells mod_dingaling that it should only pick candidates that pass the acl list given the one we made called wan excludes all the private ranges. If you update to latest trunk this list is created internally as "wan.auto" so you can use that instead of making one in your config. On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > I am trying to call from my corporate network (firewalled) using Gtalk > to Freeswitch. I am not getting any audio. > > In the logs I see that mod_dingaling is using my internal corporate IP > address which is not publically addressable. > > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates > 146.xx.xx.xx:50320 > 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable > Candidate 146.xx.xx.xx:50320 > > Further on in the log, I can see GTalk sending a new candidate IP > address to use: > 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 > name=rtp > type=local > protocol=udp > username=e+JTkVHT1xEkqXGD > password=fAxU6Pr1oF9Zq48U > address=192.168.1.102 > port=50322 > pref=1.00 > > 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked > an IP [146.xx.xx.xx] > > and > > 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 > name=rtp > type=stun > protocol=udp > username=RBqyF2XNMYLfJNoU > password=DQMjon1fSVoJIRTp > address=124.xxx.xxx.xxx > port=50323 > pref=0.90 > > 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked > an IP [146.xx.xx.xx] > and > > 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 > name=rtp > type=relay > protocol=udp > username=62L5zs2FHbcUdeCJ > password=KxmNgkUmZsLfuX6S > address=209.xx.xxx.xxx > port=19295 > pref=0.50 > > 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing > session for 4085152502 > 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked > an IP [146.xx.xx.xx] > > Because of this, I never get audio. Any ideas how to fix this? > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/dbf5783c/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 23 06:49:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 08:49:23 -0500 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: References: Message-ID: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> play with it from the cli freeswitch>global_setvar foo=12345 API CALL [global_setvar(foo=12345)] output: +OK freeswitch> eval ${foo:2:1} API CALL [eval(${foo:2:1})] output: 3 freeswitch> eval ${foo:2:3} API CALL [eval(${foo:2:3})] output: 345 freeswitch> eval ${foo:3:2} API CALL [eval(${foo:3:2})] output: 45 freeswitch> eval ${foo:-4:4} API CALL [eval(${foo:-4:4})] output: 2345 On Tue, Jun 23, 2009 at 8:33 AM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > Hello, > I once found in the wiki a page explaining how to "substring" a channel > variable, > something like > <@[intra]lanman> 12345 would be 345 if you do ${var:2} > > I can't find that page on the wiki anymore, any hint on were it could > be? :-) > > Also do you think it could be useful to extend this functionality with a > sort of Java indexOf() to extract a specific substring from a variable > (but without knowing its size like in the example above)? > > Regards, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/237e5c1a/attachment-0002.html From brian at freeswitch.org Tue Jun 23 06:23:31 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 08:23:31 -0500 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Message-ID: <41DAEAFC-5EF1-4637-8CBE-92CDC9F83E72@freeswitch.org> No it snot because of this.. you have to understand how Jingle works and if you notice it has three candidates.... 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Its already picked this one, maybe a packet capture would clear this up. /b On Jun 23, 2009, at 7:51 AM, Mark Campbell-Smith wrote: > Because of this, I never get audio. Any ideas how to fix this? > > Thanks! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/2b8f1d3c/attachment-0002.html From max.bridgewater at gmail.com Tue Jun 23 07:04:25 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 23 Jun 2009 10:04:25 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> References: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> Message-ID: Hi Michael, Using loopback solves my problem. Thanks a lot. There is a strange thing i observed though. I need to paste my extension in the default.xml file. Having them in the default directory isn't enough. Is that normal? Max. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/307fd05f/attachment-0002.html From msc at freeswitch.org Tue Jun 23 07:44:51 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 23 Jun 2009 07:44:51 -0700 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <87f2f3b90906191355v53294ff3jc4f7e5aebae00710@mail.gmail.com> <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> Message-ID: <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> On Jun 23, 2009, at 7:04 AM, Max Bridgewater wrote: > > Hi Michael, > > Using loopback solves my problem. Thanks a lot. > There is a strange thing i observed though. I need to paste my > extension in the default.xml file. Having them in the default > directory isn't enough. Is that normal? > No it isn't. What is the name of the file that has your extension and what subdir is it in? Can you pb the contents? -MC > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From max.bridgewater at gmail.com Tue Jun 23 07:59:56 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 23 Jun 2009 10:59:56 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> References: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> Message-ID: The file is located under /usr/local/freeswitch/conf/dialplan/default/. The name is: mysocket.xml. The content is: Max. On Tue, Jun 23, 2009 at 10:44 AM, Michael S Collins wrote: > > On Jun 23, 2009, at 7:04 AM, Max Bridgewater > wrote: > > > > > Hi Michael, > > > > Using loopback solves my problem. Thanks a lot. > > There is a strange thing i observed though. I need to paste my > > extension in the default.xml file. Having them in the default > > directory isn't enough. Is that normal? > > > > No it isn't. What is the name of the file that has your extension and > what subdir is it in? Can you pb the contents? > -MC > > > Max. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/52e11a9b/attachment-0002.html From mike at jerris.com Tue Jun 23 08:24:59 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 11:24:59 -0400 Subject: [Freeswitch-users] How to enable compact SIP headers in mod_sofia In-Reply-To: References: Message-ID: If you can supply a patch to expose this as a config option for us it would be appreciated. Patches can be posted to http://jira.freeswitch.org . Mike On Jun 17, 2009, at 3:22 PM, Muhammad Shahzad wrote: > Ok, thanks, i will take care of it in my code where necessary. > > Thank you. > > > On Thu, Jun 18, 2009 at 12:54 AM, Brian West > wrote: > Its not possible right now but you could if you enable the config > option and apply the tag... its something I have thought about adding > but wasn't high on my list. > > NTATAG_SIPFLAGS(MSG_FLG_COMPACT) > > http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 > > /b > > On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: > > > Hi, > > > > Is it possible to enable compact SIP headers in mod_sofia > > configuration? If yes, then how to do so? Kindly give an example. > > > > Thank you. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/0e128c86/attachment-0002.html From rupa at rupa.com Tue Jun 23 08:38:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 23 Jun 2009 10:38:51 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <005401c9f406$3ced3710$b6c7a530$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> Message-ID: How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg > and mac/phone specific cfg? settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns > *Sent:* Monday, June 22, 2009 2:57 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Sounds like a config issue in the tag. Check global sip.cfg and > mac/phone specific cfg. When you are dialing on-hook I don't think it will > use your .digitmap or ..digitmap.timer settings. When you dial off-hook it > sure will. > > On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines > on the phone. The first two are registered with a SwitchVox, the last with > Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6ae9eac9/attachment-0002.html From apt.get at gmail.com Tue Jun 23 09:01:30 2009 From: apt.get at gmail.com (David Burgess) Date: Tue, 23 Jun 2009 10:01:30 -0600 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <4A404CAA.5080809@isp-systems.net> References: <4A404CAA.5080809@isp-systems.net> Message-ID: On Mon, Jun 22, 2009 at 9:31 PM, murrah boswell wrote: > Hello All, > > I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance > on how to setup a testbed in a thin client environment. I think this would be a fairly simple matter of installing freeswitch and your softphone of choice on your ltsp server and configuring your extensions to register on 127.0.0.1, or whatever interface your freeswitch internal profile will be active on. In other words, ltsp is designed such that you can install your telephony on the server and the ltsp infrastructure will proliferate that functionality to your thin client. Install freeswitch and ekiga on the server and get ekiga to register. If you have trouble with that then this would be the place to ask. Once you get your ekiga extension registered and you are able to call voice mail, moh, etc, then log into a thin client and try the same from there. I think it will just work, but if not, that would be a good problem for the ltsp-discuss mailing list. https://lists.sourceforge.net/lists/listinfo/ltsp-discuss db From mike at jerris.com Tue Jun 23 09:11:16 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:11:16 -0400 Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <24109532.post@talk.nabble.com> References: <24109532.post@talk.nabble.com> Message-ID: <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> Try turning up your logging level to debug to see why the call is hanging up. Mike On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: > > My freeswitch has a mysql database consists of freeswitch tables, > registrations and nibblebill on mysql configured it correctly and > working... > Issue is when I call external ip's sometimes it works sometimes not? > > 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 > switch_core_session_enable_heartbeat() sofia/internal/ > 1006 at 116.5.231.40 > setting session heartbeat to 1 second(s). > 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 > switch_core_standard_on_execute() Hangup sofia/internal/1006 at 116.50.231.72 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40 > ) > Ended > 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/internal/1006 at 116.5.231.40 > [CS_DESTROY] > > On my acl.conf.xml I allow ip 116.5.231.40 > > > > > > > > I put this on my external and internal profile > > param name="apply-inbound-acl" value="globals"/> > > And put auth-calls to false... > > Please help me am really near to my success here in freeswitch... > Thanks... From mike at jerris.com Tue Jun 23 09:20:06 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:20:06 -0400 Subject: [Freeswitch-users] channel variable sip_to_tag In-Reply-To: <4A3E0607.1060608@xpirio.com> References: <4A3E0607.1060608@xpirio.com> Message-ID: if you need to use the same tags, we should be using the whole same nh in the code. There is code to do this by call uuid but I can't recall if thats for NOTIFY or INFO. If its the wrong one, we should add teh same for what you need. Mike On Jun 21, 2009, at 6:05 AM, Christian L?schenkohl wrote: > hello > > do someone know how to get the sip_to_tag from an active call? > the sip_from_tag is available as a channel variable but sip_to_tag > isn't. > i don't know if it is available at call setup, the fist time i see > the tag=... > in the sip header is the challenge response answer from fs > > i need this to get my aoc (advice-of-charge) implementation running, > this one > is based on sip info messages and has to contain the same tag's as > the active call. > > br > > -- > Ing. Christian L?schenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl at xpirio.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 23 09:26:30 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:26:30 -0400 Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 In-Reply-To: <004e01c9f378$c1707e90$44517bb0$@net> References: <000001c9ed45$72058320$56108960$@net> <00cd01c9f2a9$ad1ec700$075c5500$@net> <004e01c9f378$c1707e90$44517bb0$@net> Message-ID: <1257CF28-EE67-4E9F-8B1A-1A25F1A7C27A@jerris.com> if you run from the visual studio ide, there is a windows that shows the stack trace. Mike On Jun 22, 2009, at 4:33 PM, Drago Totev wrote: > Michael, > > It is definitely a problem in this build (13754M) (Windows) > > I found build from March and it does not brakes (still no joy, but > not sure for now if the exchange side is configured properly). > > Again, I would submit bug report is you can point me to instructions > how to do collect backtrace on Windows OS. > > Drago > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Jerris > Sent: Sunday, June 21, 2009 4:31 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSwitch and Exchange 2007 > > If this is still the case with current svn trunk, please get a > backtrace and post it to http://jira.freeswitch.org > > On Jun 21, 2009, at 3:51 PM, "Drago Totev" > wrote: > After days of running around clueless, I have no other option but to > ask the community for help one last time? > > Here is what happens: > > 1. FS sends INVITE > > 2. Exchange answers with ?302 Moved Temporarily? > > 3. FS bombs and closes > > > > Request-Line: INVITE sip:4783874764 at 10.0.0.71;transport=tcp SIP/ > 2.0 > Message Header > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > Max-Forwards: 68 > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > To: > SIP to address: sip:4783874764 at 10.0.0.71 > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > CSeq: 116688296 INVITE > Contact: > Contact Binding: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13754M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Privacy: none > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 236 > P-Asserted-Identity: "MILLEDGEVL GA" > > Message Body > > > > Status-Line: SIP/2.0 100 Trying > Message Header > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > TO: > SIP to address: sip:4783874764 at 10.0.0.71 > CSEQ: 116688296 INVITE > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > CONTENT-LENGTH: 0 > > > Status-Line: SIP/2.0 302 Moved Temporarily > Message Header > FROM: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > TO: ;tag=27df6afe0 > SIP to address: sip:4783874764 at 10.0.0.71 > SIP tag: 27df6afe0 > CSEQ: 116688296 INVITE > CALL-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > VIA: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > CONTACT: > Contact Binding: > > CONTENT-LENGTH: 0 > SERVER: RTCC/3.0.0.0 > > > Request-Line: ACK sip:4783874764 at 10.0.0.71;transport=tcp SIP/2.0 > Message Header > Via: SIP/2.0/TCP 10.8.4.3;branch=z9hG4bKQcm1vS26UU3KF > Max-Forwards: 68 > From: "MILLEDGEVL GA" 4782512197 at 10.8.4.3>;tag=Qr4j0XX18XD1m > SIP Display info: "MILLEDGEVL GA" > SIP from address: sip:4782512197 at 10.8.4.3 > SIP tag: Qr4j0XX18XD1m > To: ;tag=27df6afe0 > SIP to address: sip:4783874764 at 10.0.0.71 > SIP tag: 27df6afe0 > Call-ID: 4cdf31d7-d93e-122c-9f93-11adca034915 > CSeq: 116688296 ACK > Content-Length: 0 > > > Here is the FS log beginning the the processing of the call: > > 2009-06-21 15:49:18.636018 [NOTICE] switch_channel.c:602 New Channel sofia/external/4782512197 at 209.249.3.59 > [637f8a72-8034-254f-9937-b321c0d87414] > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3210 Channel sofia/external/4782512197 at 209.249.3.59 > entering state [received][100] > 2009-06-21 15:49:18.636018 [DEBUG] sofia.c:3217 Remote SDP: > v=0 > o=NXT02 19785 8060 IN IP4 209.249.3.59 > s=sip call > c=IN IP4 209.249.3.60 > t=0 0 > m=audio 36292 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-06-21 15:49:18.636018 [DEBUG] sofia_glue.c:3079 Audio Codec > Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:2037 Set Codec sofia/external/4782512197 at 209.249.3.59 > PCMU/8000 20 ms 160 samples > 2009-06-21 15:49:18.637018 [DEBUG] sofia_glue.c:3039 Set 2833 dtmf > payload to 101 > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_NEW > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_state_machine.c:403 (sofia/external/4782512197 at 209.249.3.59 > ) State NEW > 2009-06-21 15:49:18.637018 [DEBUG] sofia.c:3376 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_NEW -> CS_INIT > 2009-06-21 15:49:18.637018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_INIT > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59 > ) State INIT > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:83 sofia/external/4782512197 at 209.249.3.59 > SOFIA INIT > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:111 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_INIT -> CS_ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:480 (sofia/external/4782512197 at 209.249.3.59 > ) State INIT going to sleep > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59 > ) State ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:130 sofia/external/4782512197 at 209.249.3.59 > SOFIA ROUTING > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:78 sofia/external/4782512197 at 209.249.3.59 > Standard ROUTING > 2009-06-21 15:49:18.647018 [INFO] mod_dialplan_xml.c:252 Processing > MILLEDGEVL GA->4783874190 in context public > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >unloop] continue=false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >outside_call] continue=true > Dialplan: sofia/external/4782512197 at 209.249.3.59 Absolute Condition > [outside_call] > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > set(outside_call=true) > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >call_debug] continue=true > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [call_debug] ${call_debug}(true) =~ /^true$/ break=never > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action info() > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >public_extensions] continue=false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (FAIL) > [public_extensions] destination_number(4783874190) =~ /^(10[01][0-9]) > $/ break=on-false > Dialplan: sofia/external/4782512197 at 209.249.3.59 parsing [public- > >Local_UM] continue=false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Regex (PASS) > [Local_UM] ${sip_to_user}(4783874764) =~ /^(4783874764)$/ break=on- > false > Dialplan: sofia/external/4782512197 at 209.249.3.59 Action > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:114 (sofia/external/4782512197 at 209.249.3.59 > ) State Change CS_ROUTING -> CS_EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_session.c:933 Send > signal sofia/external/4782512197 at 209.249.3.59 [BREAK] > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:483 (sofia/external/4782512197 at 209.249.3.59 > ) State ROUTING going to sleep > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:397 (sofia/external/4782512197 at 209.249.3.59 > ) Running State Change CS_EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:490 (sofia/external/4782512197 at 209.249.3.59 > ) State EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] mod_sofia.c:173 sofia/external/4782512197 at 209.249.3.59 > SOFIA EXECUTE > 2009-06-21 15:49:18.647018 [DEBUG] switch_core_state_machine.c:151 sofia/external/4782512197 at 209.249.3.59 > Standard EXECUTE > EXECUTE sofia/external/4782512197 at 209.249.3.59 set(outside_call=true) > 2009-06-21 15:49:18.647018 [DEBUG] mod_dptools.c:748 sofia/external/4782512197 at 209.249.3.59 > SET [outside_call]=[true] > EXECUTE sofia/external/4782512197 at 209.249.3.59 info() > 2009-06-21 15:49:18.648018 [INFO] mod_dptools.c:946 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Channel-Read-Codec-Name: [PCMU] > Channel-Read-Codec-Rate: [8000] > Channel-Write-Codec-Name: [PCMU] > Channel-Write-Codec-Rate: [8000] > Caller-Username: [4782512197] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [MILLEDGEVL GA] > Caller-Caller-ID-Number: [4782512197] > Caller-Network-Addr: [209.249.3.59] > Caller-Destination-Number: [4783874190] > Caller-Unique-ID: [637f8a72-8034-254f-9937-b321c0d87414] > Caller-Source: [mod_sofia] > Caller-Context: [public] > Caller-Channel-Name: [sofia/external/4782512197 at 209.249.3.59] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1245613758636018] > Caller-Channel-Created-Time: [1245613758636018] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [209.249.3.59] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [4782512197] > variable_sip_from_uri: [4782512197 at 209.249.3.59] > variable_sip_from_host: [209.249.3.59] > variable_sip_from_user_stripped: [4782512197] > variable_sip_from_tag: [3454602382-411732] > variable_sofia_profile_name: [external] > variable_sip_cid_type: [pid] > variable_sip_req_user: [gw+Broadvox] > variable_sip_req_port: [5080] > variable_sip_req_uri: [gw+Broadvox at 71.29.0.61:5080] > variable_sip_req_host: [71.29.0.61] > variable_sip_to_user: [4783874764] > variable_sip_to_port: [5060] > variable_sip_to_uri: [4783874764 at 209.249.3.56:5060] > variable_sip_to_host: [209.249.3.56] > variable_sip_contact_user: [4782512197] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [4782512197 at 209.249.3.59:5060] > variable_sip_contact_host: [209.249.3.59] > variable_channel_name: [sofia/external/4782512197 at 209.249.3.59] > variable_sip_call_id: [6891536-3454602382-411724 at NXT02.broadvox.net] > variable_sip_via_host: [209.249.3.59] > variable_sip_via_port: [5060] > variable_max_forwards: [69] > variable_sip_call_info: [ 209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"] > variable_sip_gateway: [Broadvox] > variable_switch_r_sdp: [v=0 > o=NXT02 19785 8060 IN IP4 209.249.3.59 > s=sip call > c=IN IP4 209.249.3.60 > t=0 0 > m=audio 36292 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ] > variable_remote_media_ip: [209.249.3.60] > variable_remote_media_port: [36292] > variable_read_codec: [PCMU] > variable_read_rate: [8000] > variable_write_codec: [PCMU] > variable_write_rate: [8000] > variable_endpoint_disposition: [RECEIVED] > variable_outside_call: [true] > variable_current_application: [info] > > > EXECUTE sofia/external/4782512197 at 209.249.3.59 > bridge({absolute_codec_string=PCMA}sofia/gateway/um.gmc.cc.ga.us/ > 4783874764) > 2009-06-21 15:49:18.648018 [DEBUG] switch_ivr_originate.c:1017 > variable string 0 = [absolute_codec_string=PCMA] > 2009-06-21 15:49:18.648018 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/4783874764 [34ad0b66-9324-604e-931a-d0e6468ce858] > 2009-06-21 15:49:18.648018 [DEBUG] mod_sofia.c:2737 (sofia/internal/ > 4783874764) State Change CS_NEW -> CS_INIT > 2009-06-21 15:49:18.648018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_INIT > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:83 sofia/internal/ > 4783874764 SOFIA INIT > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 4783874764) State Change CS_INIT -> CS_ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/4783874764) State INIT going to sleep > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] mod_sofia.c:130 sofia/internal/ > 4783874764 SOFIA ROUTING > 2009-06-21 15:49:18.650018 [DEBUG] switch_ivr_originate.c:63 (sofia/ > internal/4783874764) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_session.c:933 Send > signal sofia/internal/4783874764 [BREAK] > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/4783874764) State ROUTING going to sleep > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/4783874764) Running State Change CS_CONSUME_MEDIA > 2009-06-21 15:49:18.650018 [DEBUG] switch_core_state_machine.c:502 > (sofia/internal/4783874764) State CONSUME_MEDIA > 2009-06-21 15:49:18.651018 [DEBUG] sofia.c:3210 Channel sofia/ > internal/4783874764 entering state [calling][0] > > > Will trade my first born for little help J > > Drago > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Drago Totev > Sent: Sunday, June 14, 2009 7:11 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] FreeSwitch and Exchange 2007 > > Hello everyone, > > I am trying to set a test environment with FS and Exchange 2007. > > Followed the instructions listed here: http://wiki.freeswitch.org/wiki/Exchange_2007_UM > and it does not seem to work. Thus far I use the default settings, > one ATA registered and confirmed to work. > > According external source: > > ?MS Exchange does not understand 'From' fields with domain name in > the address (example: field containing > address).The ?From? field?s address to be in the format ? >?; otherwise, MS Exchange drops the call.? > > I don?t know if this is the only problem? However, I see exactly > this behavior: > > ?PBX initiates call to MS Exchange by sending a SIP INVITE message > to the 5060 port of MS Exchange. MS Exchange sends 302 (Moved > Temporarily) response to PBX asking to repeat INVITE to a different > port (5065 for example). After PBX repeats the INVITE sending, the > call is established.? > > After ?302 (Moved Temporarily?, FreeSwitch bombs and closes without > any record in the log. > > Can someone help with working configuration, please? Exchange 2007 > UM role Version: 08.01.0359.002 > > Thanks in advance. > > Drago > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/9056a56b/attachment-0002.html From Richard.Lamkin at mettoni.com Tue Jun 23 09:31:06 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 23 Jun 2009 17:31:06 +0100 Subject: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. 2 - I do not want that incoming call to be answered but just stay ringing. 3 - Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. So far I've managed 1 - I see on the incoming call on the event API 2 - I used sleep 180000 (3 mins) see rule below. 3 - failed - because the rule is executing a sleep command and I cannot break in with my redirect. ============ I have tested the following works as single DP rule. Using the fixed dial plan rule below I do get the SIP signalling I want but of course it's a redirect immediately and to a fixed destination. The redirect causes FS to send a "302 moved temporarily", and the move works. ============= Any suggestions would be gratefully received Richard Lamkin richard.lamkin at mettoni.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/802f759d/attachment-0002.html From mike at jerris.com Tue Jun 23 09:36:31 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 23 Jun 2009 12:36:31 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED In-Reply-To: <24158819.post@talk.nabble.com> References: <24143545.post@talk.nabble.com> <896297E6-FBEC-4167-897F-8F78B1BE320C@freeswitch.org> <24158819.post@talk.nabble.com> Message-ID: if you turn up the debug logs it should tell you why. On Jun 22, 2009, at 11:38 PM, Edmar Cruz wrote: > > Nope. I just want to call a mobile number with no register number. > > Brian West-3 wrote: >> >> I'm going to guess you're calling a registered user? If so replace >> the @ with % >> >> /b >> >> On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: >> >>> >>> Hi, >>> >>> API CALL [originate sofia/external/1001 at 116.50.456.212] >>> -ERR SERVICE_NOT_IMPLEMENTED >>> >>> I receiving this error i dont know y? Can u help mo on this? >>> >>> I dialing a mobile number on this sometimes it works... Sometimes it >>> destroys the call [CALL_DESTROY] >>> From brian at freeswitch.org Tue Jun 23 09:36:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 11:36:56 -0500 Subject: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> Message-ID: On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote: > Can anyone suggest a good way to do the following; > > 1 - I want to be alerted [via the event API] to a new incoming call. See below.. ie park. You should get an event via event socket you can decide what to do. > 2 - I do not want that incoming call to be answered but just stay > ringing. Can't really do it that way.. you can answer it but then you're responsible for generating ringback. And billing starts when you answer it. > 3 ? Then via the API I want to send a redirect command to push the > call off to a new destination of my choice, I do not want to use the > answer/deflect sequence. Try using park ... this way you put the call in limbo and you can send the call commands at your leisure. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park > So far I?ve managed > 1 - I see on the incoming call on the event API > 2 ? I used sleep 180000 (3 mins) see rule below. > 3 ? failed - because the rule is executing a sleep command and I > cannot break in with my redirect. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/17d60d6e/attachment-0002.html From chris at cloudtel.com Tue Jun 23 09:36:19 2009 From: chris at cloudtel.com (Chris Burns) Date: Tue, 23 Jun 2009 11:36:19 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> Message-ID: Basically read the polycom manual ... it is the polycom producing the dialtone and deciding when to dial the number you are entering, using its own dialplan and interdigit timers. On Tue, Jun 23, 2009 at 10:38 AM, Rupa Schomaker wrote: > How are you configuring your polycom? > > > On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > >> I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg >> and mac/phone specific cfg? settings. I also looked for digitmap but could >> find nothing. >> >> >> >> Can you be more specific? >> >> >> >> Thanks, Lars >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns >> *Sent:* Monday, June 22, 2009 2:57 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Polycom configuration problems? >> >> >> >> Sounds like a config issue in the tag. Check global sip.cfg >> and mac/phone specific cfg. When you are dialing on-hook I don't think it >> will use your .digitmap or ..digitmap.timer settings. When you dial off-hook >> it sure will. >> >> On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: >> >> I am having difficulty with a Polycom 501 and Freeswitch. There are 3 >> lines on the phone. The first two are registered with a SwitchVox, the last >> with Freeswitch. >> >> >> >> When I select the 3rd line and begin to press numbers, pressing the 3rd >> digit automatically causes the phone to begin to dial. It does not matter >> which three numbers I press, the 3rd one is magic. >> >> >> >> However, if I do not select a line before dialing and key a 10-digit >> number into the phone, then select the 3rd line, it dials out fine. >> >> >> >> You can see from the debug console output that Processing begins before it >> hits any dialplan, so that cannot be the problem. I must have the line >> defined incorrectly for Freeswitch. >> >> >> >> Thanks for any suggestions, Lars. >> >> >> >> PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 >> >> >> >> Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 >> i386 GNU/Linux >> >> >> >> 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected >> by acl "domains". Falling back to Digest auth. >> >> 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected >> by acl "domains". Falling back to Digest auth. >> >> 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel >> sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] >> >> 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 >> (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ >> 1001 at 192.168.10.29 entering state [received][100] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: >> >> v=0 >> >> o=- 1245682011 1245682011 IN IP4 192.168.10.101 >> >> s=Polycom IP Phone >> >> c=IN IP4 192.168.10.101 >> >> t=0 0 >> >> m=audio 2254 RTP/AVP 0 8 18 101 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:18 G729/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> >> >> 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 >> (sofia/internal/1001 at 192.168.10.29) State NEW >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[G7221:115:32000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[G7221:107:16000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[G722:9:8000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare >> [PCMU:0:8000:0]/[PCMU:0:8000:20] >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec >> sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload >> to 101 >> >> 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ >> 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT >> >> 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1001 at 192.168.10.29 [BREAK] >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 >> (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 >> (sofia/internal/1001 at 192.168.10.29) State INIT >> >> 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ >> 1001 at 192.168.10.29 SOFIA INIT >> >> 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ >> 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal >> sofia/internal/1001 at 192.168.10.29 [BREAK] >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 >> (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 >> (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 >> (sofia/internal/1001 at 192.168.10.29) State ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ >> 1001 at 192.168.10.29 SOFIA ROUTING >> >> 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 >> sofia/internal/1001 at 192.168.10.29 Standard ROUTING >> >> 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing >> 1001->323 in context default >> >> Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] >> continue=false >> >> Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> >> Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/942f5c09/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Jun 23 09:45:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 23 Jun 2009 17:45:39 +0100 Subject: [Freeswitch-users] Sound file or lua script not played under load Message-ID: Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/05fcd011/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Jun 23 09:54:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 23 Jun 2009 17:54:34 +0100 Subject: [Freeswitch-users] Sound file or lua script not played under load In-Reply-To: References: Message-ID: Hmm, Looking at console I'm seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 23 June 2009 17:46 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/f7359da4/attachment-0002.html From mattdfong at gmail.com Tue Jun 23 09:55:11 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 23 Jun 2009 09:55:11 -0700 Subject: [Freeswitch-users] Sound file or lua script not played under load In-Reply-To: References: Message-ID: <4256bf830906230955s60487187t350808c0c6075e7b@mail.gmail.com> Does the log show anything? if the lua script fails to execute it should appear in freeswitch.log On Tue, Jun 23, 2009 at 9:45 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Scratching my head on this one, under load FS is not playing an audio file, > OR and lua script is not getting executed. Not all the time, just some. > I?ve changed ulimit ?n to 99999 but no diff, and ideas where else I might > look? > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/8ed2a1dd/attachment-0002.html From larclap at yahoo.com Tue Jun 23 10:25:32 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 23 Jun 2009 10:25:32 -0700 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> Message-ID: <00f301c9f427$9cd311b0$d6793510$@com> Via a web browser. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: I'm sorry Chris, but I don't know where the look for the "global sip.cfg and mac/phone specific cfg" settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/58aa28ba/attachment-0002.html From rupa at rupa.com Tue Jun 23 10:46:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 23 Jun 2009 12:46:14 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <00f301c9f427$9cd311b0$d6793510$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> Message-ID: Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: > Via a web browser. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 23, 2009 8:39 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > How are you configuring your polycom? > > On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > > I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg > and mac/phone specific cfg? settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns > *Sent:* Monday, June 22, 2009 2:57 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Sounds like a config issue in the tag. Check global sip.cfg and > mac/phone specific cfg. When you are dialing on-hook I don't think it will > use your .digitmap or ...digitmap.timer settings. When you dial off-hook it > sure will. > > On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines > on the phone. The first two are registered with a SwitchVox, the last with > Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/3b186550/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 23 11:21:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 13:21:07 -0500 Subject: [Freeswitch-users] Sound file or lua script not played under load In-Reply-To: References: Message-ID: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hmm, > > > > Looking at console I?m seeing this, does this offer any additional clues to > anyone? > > > > 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nik > Middleton > *Sent:* 23 June 2009 17:46 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Sound file or lua script not played under > load > > > > Hi Guys, > > > > Scratching my head on this one, under load FS is not playing an audio file, > OR and lua script is not getting executed. Not all the time, just some. > I?ve changed ulimit ?n to 99999 but no diff, and ideas where else I might > look? > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/c0b083d2/attachment-0002.html From christian.loeschenkohl at xpirio.com Tue Jun 23 11:27:19 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Tue, 23 Jun 2009 20:27:19 +0200 Subject: [Freeswitch-users] channel variable sip_to_tag In-Reply-To: References: <4A3E0607.1060608@xpirio.com> Message-ID: <4A411E87.6000805@xpirio.com> hi thank you for your reply how can we procced? br On 2009-06-23 18:20, Michael Jerris wrote: > if you need to use the same tags, we should be using the whole same nh > in the code. There is code to do this by call uuid but I can't recall > if thats for NOTIFY or INFO. If its the wrong one, we should add teh > same for what you need. > > Mike > > On Jun 21, 2009, at 6:05 AM, Christian L?schenkohl wrote: > >> hello >> >> do someone know how to get the sip_to_tag from an active call? >> the sip_from_tag is available as a channel variable but sip_to_tag >> isn't. >> i don't know if it is available at call setup, the fist time i see >> the tag=... >> in the sip header is the challenge response answer from fs >> >> i need this to get my aoc (advice-of-charge) implementation running, >> this one >> is based on sip info messages and has to contain the same tag's as >> the active call. >> >> br >> >> -- >> Ing. Christian L?schenkohl >> Technische Leitung, Forschung& Entwicklung VoIP >> >> xpirio >> Telekommunikation& Service GmbH >> Lakeside B04 >> 9020 Klagenfurt >> Austria >> >> T +43 (0) 5 77 11 - 1000 >> F +43 (0) 5 77 11 - 1002 >> E christian.loeschenkohl at xpirio.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From nik.middleton at noblesolutions.co.uk Tue Jun 23 11:35:16 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 23 Jun 2009 19:35:16 +0100 Subject: [Freeswitch-users] Sound file or lua script not played underload In-Reply-To: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> References: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> Message-ID: They're reading an audio file from a ram disk. Wouldn't have thought that this would cause a problem or am I wrong. Running at around 400 concurrent calls Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 23 June 2009 19:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sound file or lua script not played underload Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton wrote: Hmm, Looking at console I'm seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 23 June 2009 17:46 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/ac6a431e/attachment-0002.html From anthony.minessale at gmail.com Tue Jun 23 11:48:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 23 Jun 2009 13:48:04 -0500 Subject: [Freeswitch-users] Sound file or lua script not played underload In-Reply-To: References: <191c3a030906231121i6c6c119qe76f70fb92b143ca@mail.gmail.com> Message-ID: <191c3a030906231148t771f590csa3cd8c46c5432b5@mail.gmail.com> the lines you pasted indicate something stuck playing local_stream (hold music) and not actually reading it. playing a file from a ram disk with 400 is for sure fine. I have done many thousand before. if you turn up your debugging do you see anything else about the box going wrong? On Tue, Jun 23, 2009 at 1:35 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > They?re reading an audio file from a ram disk. Wouldn?t have thought > that this would cause a problem or am I wrong. Running at around 400 > concurrent calls > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 23 June 2009 19:21 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sound file or lua script not played > underload > > > > Are you making many calls share a single local_stream? > This error usually means a handle open to a local_stream is not reading > from that stream source, such as if you paused during playback of a > local_stream. > They are only a real issue if you are getting them with no calls up. > > On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hmm, > > > > Looking at console I?m seeing this, does this offer any additional clues to > anyone? > > > > 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream > handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nik > Middleton > *Sent:* 23 June 2009 17:46 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Sound file or lua script not played under > load > > > > Hi Guys, > > > > Scratching my head on this one, under load FS is not playing an audio file, > OR and lua script is not getting executed. Not all the time, just some. > I?ve changed ulimit ?n to 99999 but no diff, and ideas where else I might > look? > > > > Regards, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/31b8ef56/attachment-0002.html From max.bridgewater at gmail.com Tue Jun 23 12:11:52 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 23 Jun 2009 15:11:52 -0400 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> Message-ID: Hi, I've got some news on this. When i move my extension to a different directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include element at the very sample place where the default is included, things work just as expected. That is, my default.xml now include following: Cheers, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/cf95c1cb/attachment-0002.html From msc at freeswitch.org Tue Jun 23 12:27:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 12:27:26 -0700 Subject: [Freeswitch-users] Help with Socket event again In-Reply-To: References: <3FC2DE33-9C16-4A8F-99CA-48A61B91C628@freeswitch.org> <42BD121D-BD5F-43F3-98D7-5DBC99569B4E@freeswitch.org> <87f2f3b90906221344j790ed97al3ab7ba3bb9485f74@mail.gmail.com> <2B986425-AB09-41E5-86B1-60B1354F7ACD@freeswitch.org> Message-ID: <87f2f3b90906231227s6070f941q97f37ecdb8a2e43f@mail.gmail.com> I love it when users figure it out AND report back what they did to solve the issue! Nice work. -MC On Tue, Jun 23, 2009 at 12:11 PM, Max Bridgewater wrote: > Hi, > > I've got some news on this. When i move my extension to a different > directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include > element at the very sample place where the default is included, things work > just as expected. That is, my default.xml now include following: > > > > > Cheers, > Max. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/a5ac44cf/attachment-0002.html From msc at freeswitch.org Tue Jun 23 12:35:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 12:35:50 -0700 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <191c3a030906230644i154a6198j2cf1027e688297d3@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> <191c3a030906230644i154a6198j2cf1027e688297d3@mail.gmail.com> Message-ID: <87f2f3b90906231235t382df4c5jed4c95941a68c450@mail.gmail.com> Also, if and when you get this working please send a message to the list. I'd like to make sure that your setup gets documented on the wiki. -MC On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try adding this to your jingle in client.xml > > > > then edit acl.conf.xml and add this list > > > > > > > > this tells mod_dingaling that it should only pick candidates that pass the > acl list given > the one we made called wan excludes all the private ranges. > > If you update to latest trunk this list is created internally as "wan.auto" > so you can use that > instead of making one in your config. > > > > On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith < > mcampbellsmith at gmail.com> wrote: > >> Hi! >> >> I am trying to call from my corporate network (firewalled) using Gtalk >> to Freeswitch. I am not getting any audio. >> >> In the logs I see that mod_dingaling is using my internal corporate IP >> address which is not publically addressable. >> >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates >> 146.xx.xx.xx:50320 >> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable >> Candidate 146.xx.xx.xx:50320 >> >> Further on in the log, I can see GTalk sending a new candidate IP >> address to use: >> 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 >> name=rtp >> type=local >> protocol=udp >> username=e+JTkVHT1xEkqXGD >> password=fAxU6Pr1oF9Zq48U >> address=192.168.1.102 >> port=50322 >> pref=1.00 >> >> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked >> an IP [146.xx.xx.xx] >> >> and >> >> 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 >> name=rtp >> type=stun >> protocol=udp >> username=RBqyF2XNMYLfJNoU >> password=DQMjon1fSVoJIRTp >> address=124.xxx.xxx.xxx >> port=50323 >> pref=0.90 >> >> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked >> an IP [146.xx.xx.xx] >> and >> >> 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 >> name=rtp >> type=relay >> protocol=udp >> username=62L5zs2FHbcUdeCJ >> password=KxmNgkUmZsLfuX6S >> address=209.xx.xxx.xxx >> port=19295 >> pref=0.50 >> >> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing >> session for 4085152502 >> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked >> an IP [146.xx.xx.xx] >> >> Because of this, I never get audio. Any ideas how to fix this? >> >> Thanks! >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/ebf1a027/attachment-0002.html From marketing at cluecon.com Tue Jun 23 12:19:12 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 23 Jun 2009 12:19:12 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Important Information Message-ID: <87f2f3b90906231219le7753c9p3a2940ef9b5ccb58@mail.gmail.com> I know you are all eagerly anticipating the arrival of the coolest conference around! We want to make sure that everyone is aware of the following information: * The last day to get the early-bird registration is Wednesday, July 1. Early birds get into the conference for only $499. After July 1 the price is $699 per person. Please call 877.742.CLUE and get registered today! * The last day to book a hotel room at the Wyndham is Tuesday, July 21. Be sure to use expedia.com to get the best deal available. The ClueCon team is working hard to make this a very special event and we hope to have more announcements soon. You don't want to miss ClueCon 2009 - it will be the best conference you attend this year, bar none! -The ClueCon Team http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/fe040cf9/attachment-0002.html From msc at freeswitch.org Tue Jun 23 12:55:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 12:55:07 -0700 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <4A404CAA.5080809@isp-systems.net> References: <4A404CAA.5080809@isp-systems.net> Message-ID: <87f2f3b90906231255iecb6d4fm4635290051f9cf23@mail.gmail.com> Curious - what kinds of SIP phones do the clients support? Have you decided what you'd be using? -MC On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell wrote: > Hello All, > > I am an absolute newbee in the voip world but have a project where I > believe freeswitch will work and need very, very basic guidance > on how to setup a testbed in a thin client environment. > > I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and > utilize fl_teachertool 0.07 to monitor the connected > terminal clients (TCs). If you are not familiar with fl_teachertool, it > allows a teacher to view thumbnail images of each TC logged > in to the server. The teacher can click on any thumbnail and enlarge the > view, monitor all applications running on a given TC, and > take control of the keyboard and mouse of the TC. These are just a few of > the capabilities of fl_teachertool. > > What I want to do is allow the teacher to establish voice communication > using headsets and microphones with any one of the TCs by > making a "phone" call via ethernet based upon ip of the TC through > freeswitch using a softphone. > > Does this sound like something that is possible using freeswitch? If so, > could someone please give me very basic instructions on how > to setup this proof of concept? If I can just get a "teacher" stationed at > my server talking to one "student" at a TC, I believe I > can go from there. Currently I have a voiper softphone that functions, I > believe, under gnome, but I have no idea how to configure > the voiper to initiate calls through freeswitch or how to configure > freeswitch to route the call to one of my TCs. > > I also need to keep this system fully self contained. That is, I can not > have a requirement to use an outside sip service provider. > > Also, I would use any other linux sip softphones known to work with > freeswitch that people feel would work better than a voiper. > voiper seems to be more windows and mac based. I would really like to use > an ekiga since they seem to be more linux based, but I do > not believe that they have been thoroughly tested with freeswitch. > > Any help would be greatly appreciated! > > > Regards, > Murrah Boswell > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/f1b37ba1/attachment-0002.html From msc at freeswitch.org Tue Jun 23 13:15:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 13:15:31 -0700 Subject: [Freeswitch-users] voicemail problem In-Reply-To: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> References: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> Message-ID: <87f2f3b90906231315s7e8ea782j821997fb46ba7937@mail.gmail.com> Did you ever get resolution on this? If not, join us on IRC and we'll discuss it. -MC On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > I have a problem with voicemail in that freeswitch fails to let users > leave their message. Something wrong in the config I guess. I see > this in the logs: > > 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message] (en:en) > 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU > 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-goodbye.wav] (en:en) > > I assume the vm-record_message.PCMU is the file that will be created > to record the voicemail. Is that correct and how can I fix this? > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/4c014b04/attachment-0002.html From brian at freeswitch.org Tue Jun 23 13:50:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 15:50:17 -0500 Subject: [Freeswitch-users] voicemail problem In-Reply-To: <87f2f3b90906231315s7e8ea782j821997fb46ba7937@mail.gmail.com> References: <33c87fa30906201828o5e434f56hec960767ef843b26@mail.gmail.com> <87f2f3b90906231315s7e8ea782j821997fb46ba7937@mail.gmail.com> Message-ID: You're using native files and you have no native files in PCMU... /b On Jun 23, 2009, at 3:15 PM, Michael Collins wrote: > Did you ever get resolution on this? If not, join us on IRC and > we'll discuss it. > -MC > > On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith > wrote: > Hi! > > I have a problem with voicemail in that freeswitch fails to let users > leave their message. Something wrong in the config I guess. I see > this in the logs: > > 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message] (en:en) > 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm- > record_message.PCMU > 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No > language specified - Using [en] > 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-goodbye.wav] (en:en) > > I assume the vm-record_message.PCMU is the file that will be created > to record the voicemail. Is that correct and how can I fix this? > > Thanks! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/3fc78964/attachment-0002.html From larclap at yahoo.com Tue Jun 23 14:57:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 23 Jun 2009 14:57:11 -0700 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> Message-ID: <017201c9f44d$90287740$b07965c0$@com> Thanks to Rupa and Chris for this help. I didn't know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out. Are Polycoms the only SIP phones which have this feature? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 10:46 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: Via a web browser. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: I'm sorry Chris, but I don't know where the look for the "global sip.cfg and mac/phone specific cfg" settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ...digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001 at 192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001 at 192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001 at 192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001 at 192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001 at 192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/617fc764/attachment-0002.html From brian at freeswitch.org Tue Jun 23 15:04:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 23 Jun 2009 17:04:34 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <017201c9f44d$90287740$b07965c0$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> <017201c9f44d$90287740$b07965c0$@com> Message-ID: Nope other phones have this also. /b On Jun 23, 2009, at 4:57 PM, Lars Zeb wrote: > Thanks to Rupa and Chris for this help. I didn?t know enough to > understand Chris was pointing me to the Polycom phone rather than > FS. I would never have figured this out. > > Are Polycoms the only SIP phones which have this feature? > > Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/9d1b1f43/attachment-0002.html From rupa at rupa.com Tue Jun 23 15:06:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 23 Jun 2009 17:06:03 -0500 Subject: [Freeswitch-users] Polycom configuration problems? In-Reply-To: <017201c9f44d$90287740$b07965c0$@com> References: <003f01c9f37a$e8b70d00$ba252700$@com> <005401c9f406$3ced3710$b6c7a530$@com> <00f301c9f427$9cd311b0$d6793510$@com> <017201c9f44d$90287740$b07965c0$@com> Message-ID: Every sip phone I've used has this feature. Even ATAs -- though they tend to ship with more forgiving defaults. On Tue, Jun 23, 2009 at 4:57 PM, Lars Zeb wrote: > Thanks to Rupa and Chris for this help. I didn?t know enough to > understand Chris was pointing me to the Polycom phone rather than FS. I > would never have figured this out. > > > > Are Polycoms the only SIP phones which have this feature? > > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 23, 2009 10:46 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Ok, most of us configure the polycoms via a provisioning interface. > usually ftp or http. > > Anyway, when using the web interface, you want to look at: > > Goto the web interface, Click on SIP. > > Scroll down to the Local Settings section and you need to modify digitmap > and digitmap timeout. the syntax is in the polycom manuals which you can > donwload from polycom. > > On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: > > Via a web browser. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Tuesday, June 23, 2009 8:39 AM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > How are you configuring your polycom? > > On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > > I?m sorry Chris, but I don?t know where the look for the ?global sip.cfg > and mac/phone specific cfg? settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > > > Thanks, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Burns > *Sent:* Monday, June 22, 2009 2:57 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Polycom configuration problems? > > > > Sounds like a config issue in the tag. Check global sip.cfg and > mac/phone specific cfg. When you are dialing on-hook I don't think it will > use your .digitmap or ....digitmap.timer settings. When you dial off-hook it > sure will. > > On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb wrote: > > I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines > on the phone. The first two are registered with a SwitchVox, the last with > Freeswitch. > > > > When I select the 3rd line and begin to press numbers, pressing the 3rd > digit automatically causes the phone to begin to dial. It does not matter > which three numbers I press, the 3rd one is magic. > > > > However, if I do not select a line before dialing and key a 10-digit number > into the phone, then select the 3rd line, it dials out fine. > > > > You can see from the debug console output that Processing begins before it > hits any dialplan, so that cannot be the problem. I must have the line > defined incorrectly for Freeswitch. > > > > Thanks for any suggestions, Lars. > > > > PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ > 1001 at 192.168.10.29 entering state [received][100] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: > > v=0 > > o=- 1245682011 1245682011 IN IP4 192.168.10.101 > > s=Polycom IP Phone > > c=IN IP4 192.168.10.101 > > t=0 0 > > m=audio 2254 RTP/AVP 0 8 18 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729/8000 > > a=rtpmap:101 telephone-event/8000 > > > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 > (sofia/internal/1001 at 192.168.10.29) State NEW > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:115:32000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G7221:107:16000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[G722:9:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare > [PCMU:0:8000:0]/[PCMU:0:8000:20] > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec > sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples > > 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload > to 101 > > 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT > > 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1001 at 192.168.10.29 SOFIA INIT > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal > sofia/internal/1001 at 192.168.10.29 [BREAK] > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 > (sofia/internal/1001 at 192.168.10.29) State INIT going to sleep > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 > (sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 > (sofia/internal/1001 at 192.168.10.29) State ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ > 1001 at 192.168.10.29 SOFIA ROUTING > > 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1001 at 192.168.10.29 Standard ROUTING > > 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing > 1001->323 in context default > > Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop] > continue=false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > > Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/c88e83d5/attachment-0002.html From timb0311 at hotmail.com Tue Jun 23 15:12:38 2009 From: timb0311 at hotmail.com (Tim B) Date: Tue, 23 Jun 2009 18:12:38 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: Did anyone have any suggestions on this? Just to reiterate... - 8000 is a local extension defined in the default dialplan... see http://pastebin.freeswitch.org/9450 for definition - didn't work: originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/test.tif) ... see http://pastebin.freeswitch.org/9440 for log - had to add the FS ip (192.168.10.35) to the domains acl... now it to works Is this the proper way to configure? Tim From: timb0311 at hotmail.com To: freeswitch-users at lists.freeswitch.org Subject: RE: Transmit fax locally for test Date: Mon, 22 Jun 2009 18:37:47 -0400 8000 is a local extension defined in the default dialplan. Tim > ------------------------------ > > Message: 2 > Date: Mon, 22 Jun 2009 15:05:20 -0400 > From: Brian West > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > what is 8000? is it local or is it a remote endpoint? > > /b > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > test.tif) > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > Insert movie times and more without leaving Hotmail?. See how. _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing? now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try_bing_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/e0d0be43/attachment-0002.html From Richard.Lamkin at mettoni.com Tue Jun 23 15:45:47 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 23 Jun 2009 23:45:47 +0100 Subject: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138053C74CF@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7503@nickel.mettonigroup.com> Brian, Thank you for suggesting I try PARK. I tried PARK but unfortunately it sends out a 183 with SDP which stops the originator hearing ringing (ring back). If you know of a way to park without sending a 183 that would solve my problem. Regards Richard Lamkin Richard.lamkin at mettoni.com From: Brian West [mailto:brian at freeswitch.org] Sent: 23 June 2009 17:37 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem with handling unanswered calls for amanaged redirect On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote: Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. See below.. ie park. You should get an event via event socket you can decide what to do. 2 - I do not want that incoming call to be answered but just stay ringing. Can't really do it that way.. you can answer it but then you're responsible for generating ringback. And billing starts when you answer it. 3 - Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. Try using park ... this way you put the call in limbo and you can send the call commands at your leisure. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park So far I've managed 1 - I see on the incoming call on the event API 2 - I used sleep 180000 (3 mins) see rule below. 3 - failed - because the rule is executing a sleep command and I cannot break in with my redirect. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/9c1f38f0/attachment-0002.html From msc at freeswitch.org Tue Jun 23 16:14:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 23 Jun 2009 16:14:11 -0700 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: <87f2f3b90906231614t6223f65cr64e3dc492564a83c@mail.gmail.com> Is 8000 just a dialplan extension? I'm curious about the whole 8000 at 192.168.10.35 thing. I doubt that's necessary. For kicks try something like this: originate loopback/8000 &txfax(storage/fax/test.tif) That will drop the A leg right into extension 8000. -MC On Tue, Jun 23, 2009 at 3:12 PM, Tim B wrote: > Did anyone have any suggestions on this? Just to reiterate... > > - 8000 is a local extension defined in the default dialplan... see > http://pastebin.freeswitch.org/9450 for definition > > - didn't work: originate sofia/default/8000 at 192.168.10.35&txfax(storage/fax/test.tif) ... see > http://pastebin.freeswitch.org/9440 for log > > - had to add the FS ip (192.168.10.35) to the domains acl... now it to > works > > > > > > > Is this the proper way to configure? > > > Tim > > ------------------------------ > From: timb0311 at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: RE: Transmit fax locally for test > Date: Mon, 22 Jun 2009 18:37:47 -0400 > > 8000 is a local extension defined in the default dialplan. > > Tim > > > > ------------------------------ > > > > Message: 2 > > Date: Mon, 22 Jun 2009 15:05:20 -0400 > > From: Brian West > > Subject: Re: [Freeswitch-users] Transmit fax locally for test > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > > Content-Type: text/plain; charset="us-ascii" > > > > what is 8000? is it local or is it a remote endpoint? > > > > /b > > > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > > test.tif) > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------ > Insert movie times and more without leaving Hotmail?. See how. > ------------------------------ > Microsoft brings you a new way to search the web. Try Bing? now > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6df939cc/attachment-0002.html From otrcomm at isp-systems.net Tue Jun 23 17:57:49 2009 From: otrcomm at isp-systems.net (murrah boswell) Date: Tue, 23 Jun 2009 17:57:49 -0700 Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <87f2f3b90906231255iecb6d4fm4635290051f9cf23@mail.gmail.com> References: <4A404CAA.5080809@isp-systems.net> <87f2f3b90906231255iecb6d4fm4635290051f9cf23@mail.gmail.com> Message-ID: <4A417A0D.4000209@isp-systems.net> > Curious - what kinds of SIP phones do the clients support? Have you decided > what you'd be using? > -MC I am still experimenting! I have a zoiper 2.0 installed on one of my test clients. zoiper seems to work fine, so now I am attempting to get the freeswitch/zoiper interface working. I will also try to get an ekiga working, but first the zoiper. First I have to figure out how to get freeswitch operational and will be working on that tonight! Regards, Murrah Boswell > > On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell wrote: > >> Hello All, >> >> I am an absolute newbee in the voip world but have a project where I >> believe freeswitch will work and need very, very basic guidance >> on how to setup a testbed in a thin client environment. >> >> I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and >> utilize fl_teachertool 0.07 to monitor the connected >> terminal clients (TCs). If you are not familiar with fl_teachertool, it >> allows a teacher to view thumbnail images of each TC logged >> in to the server. The teacher can click on any thumbnail and enlarge the >> view, monitor all applications running on a given TC, and >> take control of the keyboard and mouse of the TC. These are just a few of >> the capabilities of fl_teachertool. >> >> What I want to do is allow the teacher to establish voice communication >> using headsets and microphones with any one of the TCs by >> making a "phone" call via ethernet based upon ip of the TC through >> freeswitch using a softphone. >> >> Does this sound like something that is possible using freeswitch? If so, >> could someone please give me very basic instructions on how >> to setup this proof of concept? If I can just get a "teacher" stationed at >> my server talking to one "student" at a TC, I believe I >> can go from there. Currently I have a voiper softphone that functions, I >> believe, under gnome, but I have no idea how to configure >> the voiper to initiate calls through freeswitch or how to configure >> freeswitch to route the call to one of my TCs. >> >> I also need to keep this system fully self contained. That is, I can not >> have a requirement to use an outside sip service provider. >> >> Also, I would use any other linux sip softphones known to work with >> freeswitch that people feel would work better than a voiper. >> voiper seems to be more windows and mac based. I would really like to use >> an ekiga since they seem to be more linux based, but I do >> not believe that they have been thoroughly tested with freeswitch. >> >> Any help would be greatly appreciated! >> >> >> Regards, >> Murrah Boswell >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From timb0311 at hotmail.com Tue Jun 23 18:23:25 2009 From: timb0311 at hotmail.com (Tim B) Date: Tue, 23 Jun 2009 21:23:25 -0400 Subject: [Freeswitch-users] Transmit fax locally for test In-Reply-To: References: Message-ID: Yeah 8000 is just a dialplan extension. That worked... originate loopback/8000 &txfax(storage/fax/test.tif) Thanks MC. I guess the loopback bypasses all the security stuff and jumps right into the dialplan looking for a matching # condition? Tim > ------------------------------ > > Message: 3 > Date: Tue, 23 Jun 2009 16:14:11 -0700 > From: Michael Collins > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90906231614t6223f65cr64e3dc492564a83c at mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Is 8000 just a dialplan extension? I'm curious about the whole > 8000 at 192.168.10.35 thing. I doubt that's necessary. For kicks try something > like this: > > originate loopback/8000 &txfax(storage/fax/test.tif) > > That will drop the A leg right into extension 8000. > > -MC > > On Tue, Jun 23, 2009 at 3:12 PM, Tim B wrote: > > > Did anyone have any suggestions on this? Just to reiterate... > > > > - 8000 is a local extension defined in the default dialplan... see > > http://pastebin.freeswitch.org/9450 for definition > > > > - didn't work: originate sofia/default/8000 at 192.168.10.35&txfax(storage/fax/test.tif) ... see > > http://pastebin.freeswitch.org/9440 for log > > > > - had to add the FS ip (192.168.10.35) to the domains acl... now it to > > works > > > > > > > > > > > > > > Is this the proper way to configure? > > > > > > Tim > > > > ------------------------------ > > From: timb0311 at hotmail.com > > To: freeswitch-users at lists.freeswitch.org > > Subject: RE: Transmit fax locally for test > > Date: Mon, 22 Jun 2009 18:37:47 -0400 > > > > 8000 is a local extension defined in the default dialplan. > > > > Tim > > > > > > > ------------------------------ > > > > > > Message: 2 > > > Date: Mon, 22 Jun 2009 15:05:20 -0400 > > > From: Brian West > > > Subject: Re: [Freeswitch-users] Transmit fax locally for test > > > To: freeswitch-users at lists.freeswitch.org > > > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > > > Content-Type: text/plain; charset="us-ascii" > > > > > > what is 8000? is it local or is it a remote endpoint? > > > > > > /b > > > > > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > > > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > > > test.tif) > > > > > > Brian West > > > brian at freeswitch.org > > > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > ------------------------------ > > Insert movie times and more without leaving Hotmail?. See how. > > ------------------------------ > > Microsoft brings you a new way to search the web. Try Bing? now > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6df939cc/attachment.html > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 36, Issue 231 > ************************************************* _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing? now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try_bing_1x1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/cc6ad227/attachment-0002.html From darklion11 at yahoo.com Tue Jun 23 19:10:20 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 23 Jun 2009 19:10:20 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> References: <24109532.post@talk.nabble.com> <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> Message-ID: <24177512.post@talk.nabble.com> Where can i find this logs? Michael Jerris wrote: > > Try turning up your logging level to debug to see why the call is > hanging up. > > Mike > > On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: > >> >> My freeswitch has a mysql database consists of freeswitch tables, >> registrations and nibblebill on mysql configured it correctly and >> working... >> Issue is when I call external ip's sometimes it works sometimes not? >> >> 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 >> switch_core_session_enable_heartbeat() sofia/internal/ >> 1006 at 116.5.231.40 >> setting session heartbeat to 1 second(s). >> 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 >> switch_core_standard_on_execute() Hangup >> sofia/internal/1006 at 116.50.231.72 >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 >> switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40 >> ) >> Ended >> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 >> switch_core_session_thread() Close Channel >> sofia/internal/1006 at 116.5.231.40 >> [CS_DESTROY] >> >> On my acl.conf.xml I allow ip 116.5.231.40 >> >> >> >> >> >> >> >> I put this on my external and internal profile >> >> param name="apply-inbound-acl" value="globals"/> >> >> And put auth-calls to false... >> >> Please help me am really near to my success here in freeswitch... >> Thanks... > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24177512.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From vince.freeswitch at hightek.org Tue Jun 23 19:15:30 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Tue, 23 Jun 2009 21:15:30 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> Message-ID: <20090624021530.GA749@quark.hightek.org> Thanks for the response Anthony. On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote: > You are way off base in a few places, let me see if I can clarify a bit. > > Here are at least 2 pointers: > > 1) The release tarballs do not come with bootstrap because they already are > bootstrapped. > 2) FreeSWITCH does not depend on system libs so all the stuff about apr is > barking up the wrong tree. > we build our own apr and apr-utils Interesting. I do not know why I got the errors I mentioned before then until I installed the exact versions of those packages it seemed to need. > I suggest you try latest svn trunk of FS and follow the BSD build guidelines > on the WIKI since you say > it's closely compatible. Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: ==================================================== Checked out the current trunk with svn. Patched /usr/include/sys/resource.h Since Dragonfly has fixed or will be fixing this future releases I patched the system header to add RLIMIT_AS rather than patching freeswitch to use RLIMIT_VMEM. Compilation still failed but there are significant improvements. bootstrap.sh seems to have been successful this time. I seems to have worked with the bsd shell this time. I also did not have to link make to gmake. It appears to have properly called gmake when building in sub-directories when gmake was run from the top. Configure completed successfully but there were these warnings: checking dlfcn.h usability... no checking dlfcn.h presence... yes configure: WARNING: dlfcn.h: present but cannot be compiled configure: WARNING: dlfcn.h: check for missing prerequisite headers? configure: WARNING: dlfcn.h: see the Autoconf documentation configure: WARNING: dlfcn.h: section "Present But Cannot Be Compiled" configure: WARNING: dlfcn.h: proceeding with the preprocessor's result configure: WARNING: dlfcn.h: in the future, the compiler will take precedence checking for dlfcn.h... yes I do not know if this is going to cause a problem. I did not have to use the "--build=i386" option to configure this time. Compiling ========= Still lots of warnings of: warning: return makes pointer from integer without a cast Errors: It is apparently not checking return codes from make. It continues even when there are errors. Is this intentional?? su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features LTCOMPILE features.lo ... Making all in sresolv LTCOMPILE sres.lo LTCOMPILE sres_cache.lo LTCOMPILE sres_blocking.lo LTCOMPILE sresolv.lo LTCOMPILE sres_sip.lo sres_sip.c: In function `sres_sip_new': sres_sip.c:267: warning: return makes pointer from integer without a cast gmake[8]: *** [sres_sip.lo] Error 1 Making all in ipt LTCOMPILE base64.lo LTCOMPILE token64.lo LINK libipt.la ... There are about 12 errors of this nature before ending with Making all in nua LTCOMPILE nua.lo nua.c: In function `nua_create': nua.c:141: warning: return makes pointer from integer without a cast nua.c:144: warning: return makes pointer from integer without a cast gmake[9]: *** [nua.lo] Error 1 gmake[8]: *** [all] Error 2 gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. gmake[7]: *** [all-recursive] Error 1 Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 It says it has been successfully built. Apparently part of the same problem of not checking the return codes. It does not say what most of the errors are except for near the last when it says No rule to make target `iptsec/libiptsec.la' It just says "Error 1" or Error 2" which does not tell me what the problem is. From mcampbellsmith at gmail.com Tue Jun 23 19:41:54 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 24 Jun 2009 12:41:54 +1000 Subject: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio? In-Reply-To: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> References: <33c87fa30906230551n7755efc5k62746c2410c1b87f@mail.gmail.com> Message-ID: <33c87fa30906231941m14d56dear6958288a54057137@mail.gmail.com> Thanks Anthony. I am getting closer. I had to put in the 146 address, which is the firewalled address I get at work. The problem now is that when the call is bridged, I do not hear audio. 2 scenarios: 1 -> the local extension is not registered. There is two way audio - I hear the voicemail in Gtalk and I can leave a message which can then be played back. 2 -> the local extension is registered. There is no audio In my incoming dialplan I am doing this bridge: It bridges okay, the phone rings, but there is no audio. On a side note: Isn't putting the candidate-acl list a temporary measure? When I travel, I will most likely get a different internal company IP address that does not start with 146. Isn't there a smarter way for dingaling to know that there is no RTP packets being received and then modify which candidate should be used? Thanks! > On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try adding this to your jingle in client.xml >> >> >> >> then edit acl.conf.xml and add this list >> >> >> ? >> ? >> ? >> >> >> this tells mod_dingaling that it should only pick candidates that pass the >> acl list given >> the one we made called wan excludes all the private ranges. >> >> If you update to latest trunk this list is created internally as "wan.auto" >> so you can use that >> instead of making one in your config. >> >> >> >> On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith < >> mcampbellsmith at gmail.com> wrote: >> >>> Hi! >>> >>> I am trying to call from my corporate network (firewalled) using Gtalk >>> to Freeswitch. ?I am not getting any audio. >>> >>> In the logs I see that mod_dingaling is using my internal corporate IP >>> address which is not publically addressable. >>> >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates >>> 146.xx.xx.xx:50320 >>> 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable >>> Candidate 146.xx.xx.xx:50320 >>> >>> Further on in the log, I can see GTalk sending a new candidate IP >>> address to use: >>> 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 >>> name=rtp >>> type=local >>> protocol=udp >>> username=e+JTkVHT1xEkqXGD >>> password=fAxU6Pr1oF9Zq48U >>> address=192.168.1.102 >>> port=50322 >>> pref=1.00 >>> >>> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked >>> an IP [146.xx.xx.xx] >>> >>> and >>> >>> 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 >>> name=rtp >>> type=stun >>> protocol=udp >>> username=RBqyF2XNMYLfJNoU >>> password=DQMjon1fSVoJIRTp >>> address=124.xxx.xxx.xxx >>> port=50323 >>> pref=0.90 >>> >>> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked >>> an IP [146.xx.xx.xx] >>> and >>> >>> 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 >>> name=rtp >>> type=relay >>> protocol=udp >>> username=62L5zs2FHbcUdeCJ >>> password=KxmNgkUmZsLfuX6S >>> address=209.xx.xxx.xxx >>> port=19295 >>> pref=0.50 >>> >>> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing >>> session for 4085152502 >>> 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked >>> an IP [146.xx.xx.xx] >>> >>> Because of this, I never get audio. ?Any ideas how to fix this? >>> >>> Thanks! From andrew at hijacked.us Tue Jun 23 20:19:51 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 23 Jun 2009 23:19:51 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090624021530.GA749@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> Message-ID: <20090624031950.GD2623@hijacked.us> On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: > Ok. I did this. > > Compilation still failed but there are significant improvements since > the last time. > > Here is what I did and the results: > It looks like some the games that sofia plays with errno makes Dragonfly unhappy. I also noticed that where the code checks for BSD-like systems (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is omitted, so obviously one of the first steps would be to fix that (if applicable). If you disable mod_sofia in modules conf, do the rest of the default modules build OK? For the record, DragonFly and FreeBSD have rather seriously diverged at this point, DragonFly forked from FreeBSD back in the 4.10 days or so and has changed a *lot* of things since, so I don't think it's gonna be quite as easy as you expected (but it's far from impossible either). Andrew From mcampbellsmith at gmail.com Tue Jun 23 20:30:42 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 24 Jun 2009 13:30:42 +1000 Subject: [Freeswitch-users] email core dump Message-ID: <33c87fa30906232030w679a24a7r39516ac7fa74572b@mail.gmail.com> Thanks Brian, but still no luck with the email.. I have configured exim4 so that I can send messages from the command line using 'mail' command and these are sent successfully. I still get a core dump in the log when freeswitch is trying to send the mail: /bin/cat: write error: Broken pipe sh: line 1: 4492 Done(1) /bin/cat /tmp/mail.1245811149abdc 4493 Segmentation fault (core dumped) | /usr/local/bin/eximcompat.sh -t xxx at xx.com 2009-06-24 12:39:09.285351 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.1245811149abdc] to [xxx at xx.com] 2009-06-24 12:39:09.285351 [DEBUG] mod_voicemail.c:2491 Sending message to xxx at xx.com eximcompat.sh is as described on the wiki: freeswitch:/# cat /usr/local/bin/eximcompat.sh #!/bin/bash exec exim4 -t Any other thoughts? From: Brian West > Subject: Re: [Freeswitch-users] email core dump > To: freeswitch-users at lists.freeswitch.org > Message-ID: <7C7A8ED9-ECED-4100-87F6-0875C054EC64 at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > /b > > On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: > > > Hi! > > > > I am trying to email from > > 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore > > original codec. > > 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to > > 1000 at 192.168.0.20 > > /bin/cat: write error: Broken pipe > > sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. > > 124558382500b1 > > 11976 Segmentation fault (core dumped) | exim4 -t > myemail at xx.com > > 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file > > [/tmp/mail.12455810042c7f] to [myemail at xx.com] > > > > I can manually send an email to myself with exim4, but freeswitch > > fails. > > > > Any ideas what I have configured incorrectly? > > > > Thanks > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/b2d38148/attachment-0002.html From mike at jerris.com Tue Jun 23 21:44:29 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Jun 2009 00:44:29 -0400 Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <24177512.post@talk.nabble.com> References: <24109532.post@talk.nabble.com> <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> <24177512.post@talk.nabble.com> Message-ID: <5ECAE2C0-82F0-4D2B-B2F3-AB86213D02A7@jerris.com> Please see the debugging pages on the wiki On Jun 23, 2009, at 10:10 PM, Edmar Cruz wrote: > > Where can i find this logs? > > Michael Jerris wrote: >> >> Try turning up your logging level to debug to see why the call is >> hanging up. >> >> Mike >> >> On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: >> >>> >>> My freeswitch has a mysql database consists of freeswitch tables, >>> registrations and nibblebill on mysql configured it correctly and >>> working... >>> Issue is when I call external ip's sometimes it works sometimes not? >>> >>> 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 >>> switch_core_session_enable_heartbeat() sofia/internal/ >>> 1006 at 116.5.231.40 >>> setting session heartbeat to 1 second(s). >>> 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 >>> switch_core_standard_on_execute() Hangup >>> sofia/internal/1006 at 116.50.231.72 >>> [CS_EXECUTE] [NORMAL_CLEARING] >>> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 >>> switch_core_session_thread() Session 7 (sofia/internal/1006 at 116.5.231.40 >>> ) >>> Ended >>> 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1006 at 116.5.231.40 >>> [CS_DESTROY] >>> >>> On my acl.conf.xml I allow ip 116.5.231.40 >>> >>> >>> >>> >>> >>> >>> >>> I put this on my external and internal profile >>> >>> param name="apply-inbound-acl" value="globals"/> >>> >>> And put auth-calls to false... >>> >>> Please help me am really near to my success here in freeswitch... >>> Thanks... >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24177512.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 23 21:53:23 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Jun 2009 00:53:23 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090624021530.GA749@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> Message-ID: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> On Jun 23, 2009, at 10:15 PM, Vincent Stemen wrote: > Thanks for the response Anthony. > > On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote: >> You are way off base in a few places, let me see if I can clarify a >> bit. >> >> Here are at least 2 pointers: >> >> 1) The release tarballs do not come with bootstrap because they >> already are >> bootstrapped. >> 2) FreeSWITCH does not depend on system libs so all the stuff about >> apr is >> barking up the wrong tree. >> we build our own apr and apr-utils > > Interesting. I do not know why I got the errors I mentioned before > then > until I installed the exact versions of those packages it seemed to > need. > > >> I suggest you try latest svn trunk of FS and follow the BSD build >> guidelines >> on the WIKI since you say >> it's closely compatible. > > Ok. I did this. > > Compilation still failed but there are significant improvements since > the last time. > > Here is what I did and the results: > > ==================================================== > Checked out the current trunk with svn. > > Patched /usr/include/sys/resource.h > > Since Dragonfly has fixed or will be fixing this future releases I > patched the > system header to add RLIMIT_AS rather than patching freeswitch to use > RLIMIT_VMEM. Can we make a patch ifdefing on RLIMIT_AS to make this always work without patches to system header files? > > > Compilation still failed but there are significant improvements. > > bootstrap.sh seems to have been successful this time. > > I seems to have worked with the bsd shell this time. > I also did not have to link make to gmake. It appears to have > properly > called gmake when building in sub-directories when gmake was run > from the top. > > Configure completed successfully but there were these warnings: > > checking dlfcn.h usability... no > checking dlfcn.h presence... yes > configure: WARNING: dlfcn.h: present but cannot be compiled > configure: WARNING: dlfcn.h: check for missing prerequisite > headers? > configure: WARNING: dlfcn.h: see the Autoconf documentation > configure: WARNING: dlfcn.h: section "Present But Cannot Be > Compiled" > configure: WARNING: dlfcn.h: proceeding with the preprocessor's > result > configure: WARNING: dlfcn.h: in the future, the compiler will take > precedence > checking for dlfcn.h... yes > This is probably fine, it means what it says, it won't try to compile with them bit the issue should probably be reported to distro maintainers > I do not know if this is going to cause a problem. > > I did not have to use the "--build=i386" option to configure this > time. > > > Compiling > ========= > > Still lots of warnings of: > warning: return makes pointer from integer without a cast > > Errors: > It is apparently not checking return codes from make. It continues > even when > there are errors. Is this intentional?? > > su_alloc.c: In function `su_salloc': > su_alloc.c:1518: warning: return makes pointer from integer without > a cast > gmake[9]: *** [su_alloc.lo] Error 1 > gmake[8]: *** [all] Error 2 > Making all in features > LTCOMPILE features.lo > ... > > Making all in sresolv > LTCOMPILE sres.lo > LTCOMPILE sres_cache.lo > LTCOMPILE sres_blocking.lo > LTCOMPILE sresolv.lo > LTCOMPILE sres_sip.lo > sres_sip.c: In function `sres_sip_new': > sres_sip.c:267: warning: return makes pointer from integer without > a cast > gmake[8]: *** [sres_sip.lo] Error 1 > Making all in ipt > LTCOMPILE base64.lo > LTCOMPILE token64.lo > LINK libipt.la > ... > > There are about 12 errors of this nature before ending with > > Making all in nua > LTCOMPILE nua.lo > nua.c: In function `nua_create': > nua.c:141: warning: return makes pointer from integer without a cast > nua.c:144: warning: return makes pointer from integer without a cast > gmake[9]: *** [nua.lo] Error 1 > gmake[8]: *** [all] Error 2 > gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed > by `libsofia-sip-ua.la'. Stop. > gmake[7]: *** [all-recursive] Error 1 > Making all in packages > gmake[6]: *** [all-recursive] Error 1 > gmake[5]: *** [all] Error 2 > gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- > ua.la] Error 2 > gmake[3]: *** [mod_sofia-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + gmake install + > +----------------------------------------------+ > gmake[1]: *** [all-recursive] Error 1 > gmake: *** [all] Error 2 > Can you post a bug to Jira.freeswitch.org with all these warnings, even better with patches to fix it. > > It says it has been successfully built. Apparently part of the same > problem of > not checking the return codes. > Patches to fix this appreciated > It does not say what most of the errors are except for near the last > when it > says > No rule to make target `iptsec/libiptsec.la' > > It just says "Error 1" or Error 2" which does not tell me what the > problem is. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dome at tel.co.th Tue Jun 23 22:36:01 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 24 Jun 2009 12:36:01 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway Message-ID: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> Dear All, Look like nibblebill does't work with multiple gatreway. I try > > > > Is 8000 just a dialplan extension? I'm curious about the whole > > 8000 at 192.168.10.35 thing. I doubt that's necessary. For kicks try > something > > like this: > > > > originate loopback/8000 &txfax(storage/fax/test.tif) > > > > That will drop the A leg right into extension 8000. > > > > -MC > > > > On Tue, Jun 23, 2009 at 3:12 PM, Tim B wrote: > > > > > Did anyone have any suggestions on this? Just to reiterate... > > > > > > - 8000 is a local extension defined in the default dialplan... see > > > http://pastebin.freeswitch.org/9450 for definition > > > > > > - didn't work: originate sofia/default/8000 at 192.168.10.35&txfax(storage/fax/test.tif) > ... see > > > http://pastebin.freeswitch.org/9440 for log > > > > > > - had to add the FS ip (192.168.10.35) to the domains acl... now it to > > > works > > > > > > > > > > > > > > > > > > > > > Is this the proper way to configure? > > > > > > > > > Tim > > > > > > ------------------------------ > > > From: timb0311 at hotmail.com > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: RE: Transmit fax locally for test > > > Date: Mon, 22 Jun 2009 18:37:47 -0400 > > > > > > 8000 is a local extension defined in the default dialplan. > > > > > > Tim > > > > > > > > > > ------------------------------ > > > > > > > > Message: 2 > > > > Date: Mon, 22 Jun 2009 15:05:20 -0400 > > > > From: Brian West > > > > Subject: Re: [Freeswitch-users] Transmit fax locally for test > > > > To: freeswitch-users at lists.freeswitch.org > > > > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67 at freeswitch.org> > > > > Content-Type: text/plain; charset="us-ascii" > > > > > > > > what is 8000? is it local or is it a remote endpoint? > > > > > > > > /b > > > > > > > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > > > > > > > > > > originate sofia/default/8000 at 192.168.10.35 &txfax(storage/fax/ > > > > > test.tif) > > > > > > > > Brian West > > > > brian at freeswitch.org > > > > > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > > > > > > > ------------------------------ > > > Insert movie times and more without leaving Hotmail?. See how.< > http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009 > > > > > ------------------------------ > > > Microsoft brings you a new way to search the web. Try Bing? now< > http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_try+bing_1x1 > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/6df939cc/attachment.html > > > > ------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > End of Freeswitch-users Digest, Vol 36, Issue 231 > > ************************************************* > > > ------------------------------ > Microsoft brings you a new way to search the web. Try Bing? now > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090623/5217ad79/attachment-0002.html From dome at tel.co.th Tue Jun 23 23:47:50 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 24 Jun 2009 13:47:50 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> Message-ID: <8ccbff060906232347i62290b26lacad4201431759f2@mail.gmail.com> 2009/6/24 Darren Schreiber > Shouldn't you be using {} not [] ? > you mean {nibble_rate=0.3}sofia/external/xxx at xxx.xxx.xxx.xxx |{nibble_rate=0.5}sofia/external/xxx at xxx.xxx.xxx.xxx I think {} use for all channel but [] for per channel Dome C. > ------------------------------ > *From:* Dome Charoenyost [mailto:dome at tel.co.th] > *Sent:* Tuesday, June 23, 2009 10:36 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Nibblebill and multiple gateway > > Dear All, > > Look like nibblebill does't work with multiple gatreway. > I try > data="nibble_account=0838833133"/> > > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx > |[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> > > nibblebill not found nibble_rate > > But > > data="nibble_account=0838833133"/> > > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx > |sofia/external/6626734000 at 202.xxx.xxx.xxx> > > Work fine > > What's difference from set application and [] ? > > Best Regards. > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/a972d7ba/attachment-0002.html From mcampbellsmith at gmail.com Wed Jun 24 00:26:47 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 24 Jun 2009 17:26:47 +1000 Subject: [Freeswitch-users] freeswitch segfault Message-ID: <33c87fa30906240026x7dc92f3aida6b963ee8b1cd81@mail.gmail.com> Hi! My call dropped and I saw this error in the syslog: Jun 24 17:05:04 freeswitch kernel: [157531.309017] freeswitch[4621]: segfault at c ip b73b2a42 sp b72a3840 error 4 in mod_sofia.so[b7369000+16c000] How can I get more information on this fault to file a bug report? Thanks! From jason at jasonjgw.net Wed Jun 24 00:47:50 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 17:47:50 +1000 Subject: [Freeswitch-users] freeswitch segfault In-Reply-To: <33c87fa30906240026x7dc92f3aida6b963ee8b1cd81@mail.gmail.com> References: <33c87fa30906240026x7dc92f3aida6b963ee8b1cd81@mail.gmail.com> Message-ID: <20090624074750.GA11226@jdc.jasonjgw.net> Mark Campbell-Smith wrote: > How can I get more information on this fault to file a bug report? See the debugging FreeSWITCH page on the wiki, and set in the FreeSWITCH core configuration (by default in switch.conf.xml), or use a ulimit -c unlimited command before running FreeSWITCH. Next time it happens, apply gdb to the core file to obtain backtraces as described on the wiki. From darklion11 at yahoo.com Wed Jun 24 01:35:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 01:35:41 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Documentation Message-ID: <24180754.post@talk.nabble.com> HI, Is there any available complete documentation for Freeswitch with matching samples aside from wiki that works. With working samples like dialplans, outbounds and prefer codecs etc. Thanks... -- View this message in context: http://www.nabble.com/Freeswitch-Documentation-tp24180754p24180754.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Wed Jun 24 01:48:53 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 18:48:53 +1000 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <24180754.post@talk.nabble.com> References: <24180754.post@talk.nabble.com> Message-ID: <20090624084853.GA18344@jdc.jasonjgw.net> Edmar Cruz wrote: > > Is there any available complete documentation for Freeswitch with matching > samples aside from wiki that works. With working samples like dialplans, > outbounds and prefer codecs etc. There isn't much besides the wiki. Most of the documentation effort has been devoted to improving the wiki rather than writing external documentation. If you would like to contribute to the wiki, you are welcome to help the community by further improving and expanding the FreeSWITCH documentation available there. From darklion11 at yahoo.com Wed Jun 24 02:10:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 02:10:13 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? Message-ID: <24181208.post@talk.nabble.com> when a type on the API of freeswitch originate sofia/external/8011104 at 116.541.23.12 1001, 8011104 my extension that I want to call and its ip is 116.541.23.12. I register on 1001 using a softphone (X-Lite) and my ip is 116.541.23.11. It works actually. but when dialing on softphone 1001 account on ip 116.541.23.11 Temporary Unavailable... What do you think is the possible issue? On originate it works sometimes but an ERR - SERVICE_NOT_IMPLEMENTED Here is my dialplan on sip_profiles/external/myprofile.xml Set acl perfectly. I set auth-calls to false. No found error on logs. Just destroying the call... Please help me... Thanks -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181208.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Wed Jun 24 02:40:42 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Wed, 24 Jun 2009 17:40:42 +0800 Subject: [Freeswitch-users] mod_dingaling no audio Message-ID: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> Hi Guys, Here's my situation: The freeswitch server and my machine are behind the same LAN. If I commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by *originate dingaling/gmail.com/userAAA at gmail.com &echo*). However, external calls have no sound at all no matter whether this param is commented out or not. Please kindly let me know what other params to set to resolve this issue. Thanks, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/afc51acb/attachment-0002.html From jason at jasonjgw.net Wed Jun 24 02:45:17 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 19:45:17 +1000 Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24181208.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> Message-ID: <20090624094517.GA22279@jdc.jasonjgw.net> Edmar Cruz wrote: > Here is my dialplan on sip_profiles/external/myprofile.xml > > > > > References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> Message-ID: search wiki from sth. like disable_rtp_autoajust , I don't remember the exact var. On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: > Hi Guys, > > Here's my situation: > > The freeswitch server and my machine are behind the same LAN. If I > commented out "ext-rtp-ip" from client.xml, I'm able to hear the > echo (by originate dingaling/gmail.com/userAAA at gmail.com &echo). > > However, external calls have no sound at all no matter whether this > param is commented out or not. > > Please kindly let me know what other params to set to resolve this > issue. > > Thanks, > -Jingwei > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/8a144b5c/attachment-0002.html From darklion11 at yahoo.com Wed Jun 24 03:02:21 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 03:02:21 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <20090624094517.GA22279@jdc.jasonjgw.net> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> Message-ID: <24181933.post@talk.nabble.com> Ooops.. Sorry wrong spelling... Same issue Jason White-14 wrote: > > Edmar Cruz wrote: > >> Here is my dialplan on sip_profiles/external/myprofile.xml >> >> >> >> >> > The above should be $1 not @1 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Claudio.Cavalera at italtel.it Wed Jun 24 03:35:32 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 24 Jun 2009 12:35:32 +0200 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> Message-ID: > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > play with it from the cli > freeswitch> eval ${foo:-4:4} > API CALL [eval(${foo:-4:4})] output: > 2345 Thanks anthm! This way to test directly on the cli is really nice, I didn't know of it. If I've understood well, a negative number as first parameter will make the second parameter useless/meaningless. Besides, the second parameter is useless/meaningless if set equal to 0 or lesser than 0. I still think that we could benefit from more power here in the dialplan, if I have 12346578 at domain.org I'm not able to grab out the number (of which I don't know the length) even if I know that domain.org is always the same length. I've looked for something useful in http://apr.apache.org/docs/apr/1.3/apr__strings_8h.html and switch_apr.c but I think I'll end up calling an external script. :) Best Regars, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From saeedahmad1981 at gmail.com Wed Jun 24 03:53:30 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 24 Jun 2009 12:53:30 +0200 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: References: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> Message-ID: <2200BAECC6FD423784C068B2E0595BA7@saeedlaptop> Can we also test dialplan using CLI, like "dial" in asterisk? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cavalera Claudio Luigi Sent: Wednesday, June 24, 2009 12:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Variable manipulation in the dialplan > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > play with it from the cli > freeswitch> eval ${foo:-4:4} > API CALL [eval(${foo:-4:4})] output: > 2345 Thanks anthm! This way to test directly on the cli is really nice, I didn't know of it. If I've understood well, a negative number as first parameter will make the second parameter useless/meaningless. Besides, the second parameter is useless/meaningless if set equal to 0 or lesser than 0. I still think that we could benefit from more power here in the dialplan, if I have 12346578 at domain.org I'm not able to grab out the number (of which I don't know the length) even if I know that domain.org is always the same length. I've looked for something useful in http://apr.apache.org/docs/apr/1.3/apr__strings_8h.html and switch_apr.c but I think I'll end up calling an external script. :) Best Regars, Claudio Internet Email Confidentiality Footer ---------------------------------------------------------------------------- ------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ---------------------------------------------------------------------------- ------------------------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jason at jasonjgw.net Wed Jun 24 04:04:13 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 24 Jun 2009 21:04:13 +1000 Subject: [Freeswitch-users] Variable manipulation in the dialplan In-Reply-To: <2200BAECC6FD423784C068B2E0595BA7@saeedlaptop> References: <191c3a030906230649m41f59dc9k7561dfed4e11fc81@mail.gmail.com> <2200BAECC6FD423784C068B2E0595BA7@saeedlaptop> Message-ID: <20090624110413.GA27336@jdc.jasonjgw.net> Saeed Ahmed wrote: > Can we also test dialplan using CLI, like "dial" in asterisk? Have a look at the originate command. From dftoro at yahoo.com Wed Jun 24 06:07:16 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 24 Jun 2009 06:07:16 -0700 (PDT) Subject: [Freeswitch-users] transfer_ringback from mod_managed Message-ID: <413467.8366.qm@web33501.mail.mud.yahoo.com> Greetings ? When I use Session.SetVariable("transfer_ringback", "us-ring") from managed code the bridge fails with "NO_ANSWER" cause. If I use from xml dial plan the call is stablished. ? I have FS rev 13750 running on Windows. ? This is a issue or I don't use properly transfer_ringback variable ? ? Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/562d4c7b/attachment-0002.html From brian at freeswitch.org Wed Jun 24 06:32:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 24 Jun 2009 08:32:00 -0500 Subject: [Freeswitch-users] transfer_ringback from mod_managed In-Reply-To: <413467.8366.qm@web33501.mail.mud.yahoo.com> References: <413467.8366.qm@web33501.mail.mud.yahoo.com> Message-ID: Well its the same you use ${us-ring} in both cases. /b On Jun 24, 2009, at 8:07 AM, Diego Toro wrote: > Greetings > > When I use Session.SetVariable("transfer_ringback", "us-ring") from > managed code the bridge fails with "NO_ANSWER" cause. If I use > from > xml dial plan the call is stablished. > > I have FS rev 13750 running on Windows. > > This is a issue or I don't use properly transfer_ringback variable ? > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/6406a6ff/attachment-0002.html From dftoro at yahoo.com Wed Jun 24 07:20:52 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 24 Jun 2009 07:20:52 -0700 (PDT) Subject: [Freeswitch-users] transfer_ringback from mod_managed Message-ID: <936224.63343.qm@web33503.mail.mud.yahoo.com> Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring}); I have message: "[CRIT] switch_channel.c:633 Invalid data (${transfer_ringback} contains a variable)". ? Using from managed code: ? string stUsRing = _Session.GetVariable("us-ring"); Session.SetVariable("ringback", stUsRing); Session.SetVariable("transfer_ringback", stUsRing); ? The bridge works fine. ? The question is, using Session is not possible acces directly global vars way ${var_name} ? ? Thanks ? Diego --- On Wed, 6/24/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] transfer_ringback from mod_managed To: freeswitch-users at lists.freeswitch.org Date: Wednesday, June 24, 2009, 8:32 AM Well its the same you use ${us-ring} in both cases. /b On Jun 24, 2009, at 8:07 AM, Diego Toro wrote: Greetings ? When I use Session.SetVariable("transfer_ringback", "us-ring") from managed code the bridge fails with "NO_ANSWER" cause. If I use from xml dial plan the call is stablished. ? I have FS rev 13750 running on Windows. ? This is a issue or I don't use properly transfer_ringback variable ? ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/5c35cd9e/attachment-0002.html From brian at freeswitch.org Wed Jun 24 07:33:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 24 Jun 2009 09:33:04 -0500 Subject: [Freeswitch-users] transfer_ringback from mod_managed In-Reply-To: <936224.63343.qm@web33503.mail.mud.yahoo.com> References: <936224.63343.qm@web33503.mail.mud.yahoo.com> Message-ID: <775DF765-03C2-4F49-BD23-E2E9BEB2085A@freeswitch.org> Chances are you need to get var us-ring then use that to set the transfer_ringback /b On Jun 24, 2009, at 9:20 AM, Diego Toro wrote: > Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring}); > I have message: "[CRIT] switch_channel.c:633 Invalid data ($ > {transfer_ringback} contains a variable)". > > Using from managed code: > > string stUsRing = _Session.GetVariable("us-ring"); > Session.SetVariable("ringback", stUsRing); > Session.SetVariable("transfer_ringback", stUsRing); > > The bridge works fine. > > The question is, using Session is not possible acces directly global > vars way ${var_name} ? > > Thanks > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/8a8fb06f/attachment-0002.html From mike at jerris.com Wed Jun 24 08:15:01 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Jun 2009 11:15:01 -0400 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> Message-ID: <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> try adding before the bridge and report back results. Mike On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: > Dear All, > > Look like nibblebill does't work with multiple gatreway. > I try > > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx > |[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> > > nibblebill not found nibble_rate > > But > > > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx > |sofia/external/6626734000 at 202.xxx.xxx.xxx> > > Work fine > > What's difference from set application and [] ? > > Best Regards. > Dome C. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/561334f5/attachment-0002.html From Richard.Lamkin at mettoni.com Wed Jun 24 08:29:34 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Wed, 24 Jun 2009 16:29:34 +0100 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg call-command: execute execute-app-name: redirect execute-app-arg: sip:@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg call-command: execute execute-app-name: hangup execute-app-arg: Using command "api show channels" I find the following held on FS The only way I've found to remove these calls is "api hupall" ------------------------- uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. ------------------- The SIP signalling is correct with an outgoing "302 moved temporarily" [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either "api hupall", or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to "api hupall". "api hupall" kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lamkin at mettoni.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/47300618/attachment-0002.html From msc at freeswitch.org Wed Jun 24 08:57:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Jun 2009 08:57:24 -0700 Subject: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips? In-Reply-To: <24177512.post@talk.nabble.com> References: <24109532.post@talk.nabble.com> <03D5893D-4983-4FC5-87B6-FA43AE2D371F@jerris.com> <24177512.post@talk.nabble.com> Message-ID: <87f2f3b90906240857k25f38864r60b305fdbc24fdc1@mail.gmail.com> On Tue, Jun 23, 2009 at 7:10 PM, Edmar Cruz wrote: > > Where can i find this logs? > Please look at this page because it will give you a lot of information about how to collect information for debugging: http://wiki.freeswitch.org/wiki/Reporting_Bugs I recommend setting aside 20 minutes to read that page and learn how to turn on debugging, capture command line interface output, etc. If you master those basic skills it will save you (and us) a lot of time. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/01b98cbe/attachment-0002.html From freeswitch-users at digitaldan.com Wed Jun 24 09:10:04 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 24 Jun 2009 10:10:04 -0600 (MDT) Subject: [Freeswitch-users] Newbee: Need Help Thin Client Environment In-Reply-To: <4A417A0D.4000209@isp-systems.net> Message-ID: <9827933.1121245859861542.JavaMail.daniel@radio> I was thinking, for the students machines your could run the command line program linphonec in auto answer mode ( linphonec -a ). When a teacher calls one of the thin clients, the student would automatically hear and be able to speak to the teacher and not have to worry about dealing with a softphone. You could probably do that as well with freeswitch running as a client on each thin client, but linphoncc does this very well today. D- ----- Original Message ----- From: "murrah boswell" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, June 23, 2009 6:57:49 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Newbee: Need Help Thin Client Environment > Curious - what kinds of SIP phones do the clients support? Have you decided > what you'd be using? > -MC I am still experimenting! I have a zoiper 2.0 installed on one of my test clients. zoiper seems to work fine, so now I am attempting to get the freeswitch/zoiper interface working. I will also try to get an ekiga working, but first the zoiper. First I have to figure out how to get freeswitch operational and will be working on that tonight! Regards, Murrah Boswell > > On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell wrote: > >> Hello All, >> >> I am an absolute newbee in the voip world but have a project where I >> believe freeswitch will work and need very, very basic guidance >> on how to setup a testbed in a thin client environment. >> >> I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and >> utilize fl_teachertool 0.07 to monitor the connected >> terminal clients (TCs). If you are not familiar with fl_teachertool, it >> allows a teacher to view thumbnail images of each TC logged >> in to the server. The teacher can click on any thumbnail and enlarge the >> view, monitor all applications running on a given TC, and >> take control of the keyboard and mouse of the TC. These are just a few of >> the capabilities of fl_teachertool. >> >> What I want to do is allow the teacher to establish voice communication >> using headsets and microphones with any one of the TCs by >> making a "phone" call via ethernet based upon ip of the TC through >> freeswitch using a softphone. >> >> Does this sound like something that is possible using freeswitch? If so, >> could someone please give me very basic instructions on how >> to setup this proof of concept? If I can just get a "teacher" stationed at >> my server talking to one "student" at a TC, I believe I >> can go from there. Currently I have a voiper softphone that functions, I >> believe, under gnome, but I have no idea how to configure >> the voiper to initiate calls through freeswitch or how to configure >> freeswitch to route the call to one of my TCs. >> >> I also need to keep this system fully self contained. That is, I can not >> have a requirement to use an outside sip service provider. >> >> Also, I would use any other linux sip softphones known to work with >> freeswitch that people feel would work better than a voiper. >> voiper seems to be more windows and mac based. I would really like to use >> an ekiga since they seem to be more linux based, but I do >> not believe that they have been thoroughly tested with freeswitch. >> >> Any help would be greatly appreciated! >> >> >> Regards, >> Murrah Boswell >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/13ffbb0b/attachment-0002.html From Richard.Lamkin at mettoni.com Wed Jun 24 09:54:39 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Wed, 24 Jun 2009 17:54:39 +0100 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7887@nickel.mettonigroup.com> I have also observed that the cpu load goes up to 100% when only a couple of orphaned calls are left without being cleared by "api hupall". Richard Lamkin richard.lamkin at mettoni.com From: Richard Lamkin Sent: 24 June 2009 16:30 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg call-command: execute execute-app-name: redirect execute-app-arg: sip:@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg call-command: execute execute-app-name: hangup execute-app-arg: Using command "api show channels" I find the following held on FS The only way I've found to remove these calls is "api hupall" ------------------------- uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. ------------------- The SIP signalling is correct with an outgoing "302 moved temporarily" [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either "api hupall", or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to "api hupall". "api hupall" kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lamkin at mettoni.com ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/3fdf0f98/attachment-0002.html From anthony.minessale at gmail.com Wed Jun 24 10:36:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 24 Jun 2009 12:36:31 -0500 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C7887@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF46138053C7887@nickel.mettonigroup.com> Message-ID: <191c3a030906241036m7c0ff8d7x6170f2e76d00db1c@mail.gmail.com> can do the following: 1) "make current" or do a fresh checkout to make sure you build is clean. 2) try executing the app "ring_ready" with no args in place of respond 180 and see if it makes any difference. 3) clear out your logfile by stopping FS and deleting /usr/local/freeswitch/log/freeswitch.log and reproduce. Then send me the log along with the list of channels you still see stuck. report the findings to jira http://jira.freeswitch.org and let me know the ticket number. make sure all your attachments have a .txt extensions when they are text files as jira has a bug of it's own with attachments and file types. On Wed, Jun 24, 2009 at 11:54 AM, Richard Lamkin wrote: > I have also observed that the cpu load goes up to 100% when only a couple > of orphaned calls are left without being cleared by ?api hupall?. > > > > Richard Lamkin > > > > richard.lamkin at mettoni.com > > > > > > *From:* Richard Lamkin > *Sent:* 24 June 2009 16:30 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Orphaned calls left on FS after redirect off > of FS > > > > I am using the API to manage calls as they arrive at FS from a trunk > > > > I have a very simple Dial plan rule that parks the incoming call. > > > > > > > > > > > > > > > > > > Once the call is parked via the API I first send a ringing (to keep the > originator happy) > > > > sendmsg > > call-command: execute > > execute-app-name: respond > > execute-app-arg: 180 > > > > Via the API I then redirect the call on to another PSTN number back through > the same gateway > > > > sendmsg > > call-command: execute > > execute-app-name: redirect > > execute-app-arg: sip:@194.0.147.16 > > > > The redirection works well and the originator and destination are connected > correctly. > > > > But after the call has left FS I?m still left with some call debris which I > cannot clear down using > > > > sendmsg > > call-command: execute > > execute-app-name: hangup > > execute-app-arg: > > > > > > Using command ?api show channels? I find the following held on FS The > only way I?ve found to remove these calls is ?api hupall? > > > > ------------------------- > > > uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate > > 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 > 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060 > ,CS_EXECUTE,0203196599,0203196599, > > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16 > ,XML,Public,PCMU,8000,PCMU,8000 > > c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 > 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060 > ,CS_EXECUTE,0203196598,0203196598, > > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16 > ,XML,Public,PCMU,8000,PCMU,8000 > > b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 > 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060 > ,CS_EXECUTE,0203196599,0203196599, > > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16 > ,XML,Public,PCMU,8000,PCMU,8000 > > 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 > 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060 > ,CS_EXECUTE,0189728400,0189728400, > > 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16 > ,XML,Public,PCMA,8000,PCMA,8000 > > > > 4 total. > > ------------------- > > > > The SIP signalling is correct with an outgoing ?302 moved temporarily? > [with the new destination in the contact] which is then Ack?ed by the > switch. From a SIP point of view the call no longer on FS. > > The only way I?ve found to remove these phantom calls is either ?api > hupall?, or restart the Sip profile. > > > > Any suggestions on how I can remove these phantom calls without recourse to > ?api hupall?. ?api hupall? kills any incoming calls as well as the stuck > calls. > > > > Regards > > > > Richard Lamkin > > richard.lamkin at mettoni.com > > > > > > > > > > > > > > > > > > ************************************************************************* > > Please consider the environment before printing this e-mail > > ************************************************************************* > > This email and any files transmitted with it are confidential and > > intended solely for the use of the individual or entity to whom they > > are addressed. If you have received this email in error please notify > > the system manager. http://www.mettoni.com > > > > Mettoni Ltd > > Registered in England and Wales: 4485956 > > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > > ************************************************************************* > > ************************************************************************* > Please consider the environment before printing this e-mail > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/ddb11d00/attachment-0002.html From msc at freeswitch.org Wed Jun 24 11:12:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Jun 2009 11:12:34 -0700 Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24181208.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> Message-ID: <87f2f3b90906241112h65775787ge540160db7cf62f4@mail.gmail.com> Turn on debugging and capture the output. Put it in a pastebin and post the link here. We'll take a look. -MC On Wed, Jun 24, 2009 at 2:10 AM, Edmar Cruz wrote: > > when a type on the API of freeswitch originate > sofia/external/8011104 at 116.541.23.12 1001, 8011104 my extension that I > want > to call and its ip is 116.541.23.12. I register on 1001 using a softphone > (X-Lite) and my ip is 116.541.23.11. It works actually. > > but when dialing on softphone 1001 account on ip 116.541.23.11 Temporary > Unavailable... > > What do you think is the possible issue? > > On originate it works sometimes but an ERR - SERVICE_NOT_IMPLEMENTED > > Here is my dialplan on sip_profiles/external/myprofile.xml > > > > > data="sofia/external/@1 at 116.541.23.12"/> > > > > Set acl perfectly. > > I set auth-calls to false. > > No found error on logs. > > Just destroying the call... > > Please help me... > > Thanks > > -- > View this message in context: > http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181208.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/f5101644/attachment-0002.html From brian at freeswitch.org Wed Jun 24 11:20:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 24 Jun 2009 13:20:50 -0500 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> Message-ID: I have tried to reproduce this issue but haven't been able too... What SVN Rev are you on? /b On Jun 24, 2009, at 10:29 AM, Richard Lamkin wrote: > I am using the API to manage calls as they arrive at FS from a trunk > > I have a very simple Dial plan rule that parks the incoming call. > > > > > > > > > Once the call is parked via the API I first send a ringing (to keep > the originator happy) > > sendmsg > call-command: execute > execute-app-name: respond > execute-app-arg: 180 > > Via the API I then redirect the call on to another PSTN number back > through the same gateway > > sendmsg > call-command: execute > execute-app-name: redirect > execute-app-arg: sip:@194.0.147.16 > > The redirection works well and the originator and destination are > connected correctly. > > But after the call has left FS I?m still left with some call debris > which I cannot clear down using > > sendmsg > call-command: execute > execute-app-name: hangup > execute-app-arg: > > > Using command ?api show channels? I find the following held on FS > The only way I?ve found to remove these calls is ?api hupall? > > ------------------------- > uuid > ,created > ,created_epoch > ,name > ,state > ,cid_name > ,cid_num > ,ip_addr > ,dest > ,application > ,application_data > ,dialplan,context,read_codec,read_rate,write_codec,write_rate > 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16 > :5060,CS_EXECUTE,0203196599,0203196599, > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public,PCMU,8000,PCMU,8000 > c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16 > :5060,CS_EXECUTE,0203196598,0203196598, > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public,PCMU,8000,PCMU,8000 > b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16 > :5060,CS_EXECUTE,0203196599,0203196599, > 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public,PCMU,8000,PCMU,8000 > 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16 > :5060,CS_EXECUTE,0189728400,0189728400, > 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public,PCMA,8000,PCMA,8000 > > 4 total. > ------------------- > > The SIP signalling is correct with an outgoing ?302 moved > temporarily? [with the new destination in the contact] which is then > Ack?ed by the switch. From a SIP point of view the call no longer > on FS. > The only way I?ve found to remove these phantom calls is either ?api > hupall?, or restart the Sip profile. > > Any suggestions on how I can remove these phantom calls without > recourse to ?api hupall?. ?api hupall? kills any incoming calls as > well as the stuck calls. > > Regards > > Richard Lamkin > richard.lamkin at mettoni.com > > > > > > > > > ************************************************************************* > Please consider the environment before printing this e-mail > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/fa9797f7/attachment-0002.html From john at feith.com Wed Jun 24 18:12:17 2009 From: john at feith.com (John Wehle) Date: Wed, 24 Jun 2009 21:12:17 -0400 (EDT) Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking Message-ID: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> I have a lua script that originates a call: local s = freeswitch.Session ( "{ignore_early_media=true,origination_caller_id_name=" .. caller .. "}loopback/" .. destination .. "/default/XML") s:execute ("sleep", "1000") ... which works fine if a valid number is supplied. However if a invalid number is supplied, then the script hits: [ERR] switch_cpp.cpp:607 session is not initalized [ERR] freeswitch_lua.cpp:102 session is not initalized What's the recommended way to check if the session constructor was successful (i.e. the number could be dialed)? My other option is: local s = freeswitch.Session () local r = s:originate (nil, "{ignore_early_media=true,origination_caller_id_name=" .. caller .. "}loopback/" .. destination .. "/default/XML", 300) if r == 1 then stream:write ("-ERR call failed\n") return end which does handle invalid numbers however there are the minor issues such as: a) The documentation seems to strongly discourage using the originate method for some reason. b) The lua originate method seems to require timeout to be specified even though the documentation implies it's optional. c) Using this approach causes the message: [WARNING] mod_limit.c:576 USAGE: hash [insert|delete]/// which I have yet to track down. Thoughts? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From jingwei.yang at gmail.com Wed Jun 24 19:23:19 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 10:23:19 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> Message-ID: <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put "disable-rtp-auto-adjust" inside client.xml. No matter what value this parameter has (true or false), local IP is able to hear the echo but external ones still have no audio. On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: > search wiki from sth. like disable_rtp_autoajust , I don't remember the > exact var. > > On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: > > Hi Guys, > > Here's my situation: > > The freeswitch server and my machine are behind the same LAN. If I > commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by > *originate dingaling/gmail.com/userAAA at gmail.com &echo*). > > However, external calls have no sound at all no matter whether this param > is commented out or not. > > Please kindly let me know what other params to set to resolve this issue. > > Thanks, > -Jingwei > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/96fff17b/attachment-0002.html From darklion11 at yahoo.com Wed Jun 24 19:37:14 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 19:37:14 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <87f2f3b90906241112h65775787ge540160db7cf62f4@mail.gmail.com> References: <24181208.post@talk.nabble.com> <87f2f3b90906241112h65775787ge540160db7cf62f4@mail.gmail.com> Message-ID: <24196071.post@talk.nabble.com> This is on my freeswitch logs... 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 sofia_glue_tech_set_codec() Set Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_INIT 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State INIT 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at 116.541.23.12 SOFIA INIT 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> CS_ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State INIT going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State ROUTING 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 Standard ROUTING 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Edmar->639273642511 in context public Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->outside_call] continue=true Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition [outside_call] Dialplan: sofia/internal/1001 at 116.541.23.11 Action set(outside_call=true) Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->call_debug] continue=true Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->public_extensions] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_extensions] destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->public_did] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_did] destination_number(639273642511) =~ /^(5551212)$/ break=on-false 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1001 at 116.541.23.11) State Change CS_ROUTING -> CS_EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State ROUTING going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State EXECUTE 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 Standard EXECUTE EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1001 at 116.541.23.11 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/1001 at 116.541.23.11 [KILL] 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State EXECUTE going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_HANGUP 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State HANGUP 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 480 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State HANGUP going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State Change CS_HANGUP -> CS_REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11) State REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/1001 at 116.541.23.11 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11) State REPORTING going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State Change CS_REPORTING -> CS_DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11) Locked, Waiting on external entities 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11) Ended 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1001 at 116.541.23.11 [CS_DESTROY] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 116.541.23.11) State DESTROY 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 Standard DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 116.541.23.11) State DESTROY going to sleep -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24196071.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Wed Jun 24 19:52:50 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 10:52:50 +0800 Subject: [Freeswitch-users] FreeSwitch at backend Message-ID: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> Hi Folks, I understand freeSwitch is supporting a couple of languages for call controls like Lua, Javascript, Perl, Java... However, after digging into the detailed wiki pages, I found out the codes written in those languages can only be executed via the freeswitch console. I was wondering whether it's possible to run FreeSwitch at backend and have a piece of Java program (outside the freeSwitch console) to invoke freeSwitch commands? Thanks, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d71c574d/attachment-0002.html From paul.degt at gmail.com Wed Jun 24 20:13:05 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Wed, 24 Jun 2009 23:13:05 -0400 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> Message-ID: <4A42EB41.8030703@gmail.com> You can use FS socket event interface for that. See free Java lib for inbound socket event here: http://versafon.com/versafonweb/Software.jsp Jingwei Yang wrote: > Hi Folks, > > I understand freeSwitch is supporting a couple of languages for call > controls like Lua, Javascript, Perl, Java... However, after digging > into the detailed wiki pages, I found out the codes written in those > languages can only be executed via the freeswitch console. I was > wondering whether it's possible to run FreeSwitch at backend and have > a piece of Java program (outside the freeSwitch console) to invoke > freeSwitch commands? > > Thanks, > -Jingwei > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Jun 24 20:24:55 2009 From: dujinfang at gmail.com (seven) Date: Thu, 25 Jun 2009 11:24:55 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> Message-ID: uncomment ext-rtp-ip On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: > Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and > put "disable-rtp-auto-adjust" inside client.xml. No matter what > value this parameter has (true or false), local IP is able to hear > the echo but external ones still have no audio. > > On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: > search wiki from sth. like disable_rtp_autoajust , I don't remember > the exact var. > > > On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >> Hi Guys, >> >> Here's my situation: >> >> The freeswitch server and my machine are behind the same LAN. If I >> commented out "ext-rtp-ip" from client.xml, I'm able to hear the >> echo (by originate dingaling/gmail.com/userAAA at gmail.com &echo). >> >> However, external calls have no sound at all no matter whether this >> param is commented out or not. >> >> Please kindly let me know what other params to set to resolve this >> issue. >> >> Thanks, >> -Jingwei >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a9cb6649/attachment-0002.html From dujinfang at gmail.com Wed Jun 24 20:26:10 2009 From: dujinfang at gmail.com (seven) Date: Thu, 25 Jun 2009 11:26:10 +0800 Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24181933.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> <24181933.post@talk.nabble.com> Message-ID: <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> put your extension in dialplan/public.xml instead of sip_profiles/ external/myprofile.xml On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: > > Ooops.. Sorry wrong spelling... Same issue > > Jason White-14 wrote: >> >> Edmar Cruz wrote: >> >>> Here is my dialplan on sip_profiles/external/myprofile.xml >>> >>> >>> >>> >>> > >> The above should be $1 not @1 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jingwei.yang at gmail.com Wed Jun 24 20:26:46 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 11:26:46 +0800 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <4A42EB41.8030703@gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> <4A42EB41.8030703@gmail.com> Message-ID: <13529f9d0906242026w21d8f0dcla822ec94bfb6a4e6@mail.gmail.com> Hi Paul, thanks for your reply. I've give it a try. On Thu, Jun 25, 2009 at 11:13 AM, paul.degt at gmail.com wrote: > You can use FS socket event interface for that. See free Java lib for > inbound socket event here: http://versafon.com/versafonweb/Software.jsp > > Jingwei Yang wrote: > > Hi Folks, > > > > I understand freeSwitch is supporting a couple of languages for call > > controls like Lua, Javascript, Perl, Java... However, after digging > > into the detailed wiki pages, I found out the codes written in those > > languages can only be executed via the freeswitch console. I was > > wondering whether it's possible to run FreeSwitch at backend and have > > a piece of Java program (outside the freeSwitch console) to invoke > > freeSwitch commands? > > > > Thanks, > > -Jingwei > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/4034fd1a/attachment-0002.html From darklion11 at yahoo.com Wed Jun 24 20:41:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Wed, 24 Jun 2009 20:41:01 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> <24181933.post@talk.nabble.com> <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> Message-ID: <24196467.post@talk.nabble.com> If then, what bridge i shall call to? Like this? dujinfang wrote: > > put your extension in dialplan/public.xml instead of sip_profiles/ > external/myprofile.xml > > > On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: > >> >> Ooops.. Sorry wrong spelling... Same issue >> >> Jason White-14 wrote: >>> >>> Edmar Cruz wrote: >>> >>>> Here is my dialplan on sip_profiles/external/myprofile.xml >>>> >>>> >>>> >>>> >>>> >> >>> The above should be $1 not @1 >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24196467.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jingwei.yang at gmail.com Wed Jun 24 20:42:09 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 11:42:09 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> Message-ID: <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> Thanks seven. External IPs have sound echo this time with ext-rtp-ip uncommented and disable-rtp-auto-adjust=true. However, internal IP has no audio this time no matter what value disable-rtp-auto-adjust is... On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: > uncomment ext-rtp-ip > > On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: > > Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put > "disable-rtp-auto-adjust" inside client.xml. No matter what value this > parameter has (true or false), local IP is able to hear the echo but > external ones still have no audio. > > On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: > >> search wiki from sth. like disable_rtp_autoajust , I don't remember the >> exact var. >> >> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >> >> Hi Guys, >> >> Here's my situation: >> >> The freeswitch server and my machine are behind the same LAN. If I >> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >> >> However, external calls have no sound at all no matter whether this param >> is commented out or not. >> >> Please kindly let me know what other params to set to resolve this issue. >> >> Thanks, >> -Jingwei >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/3c45c8b1/attachment-0002.html From chris.chen2004 at gmail.com Wed Jun 24 20:53:33 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Wed, 24 Jun 2009 23:53:33 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> Message-ID: <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> Please provide your client.xml detail with confidential information crossout, I have gtalk client and server working properly behind the NAT. I should be able to help you. Chris On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: > Thanks seven. External IPs have sound echo this time with ext-rtp-ip > uncommented and disable-rtp-auto-adjust=true. However, internal IP has no > audio this time no matter what value disable-rtp-auto-adjust is... > > > On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: > >> uncomment ext-rtp-ip >> >> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >> >> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put >> "disable-rtp-auto-adjust" inside client.xml. No matter what value this >> parameter has (true or false), local IP is able to hear the echo but >> external ones still have no audio. >> >> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >> >>> search wiki from sth. like disable_rtp_autoajust , I don't remember the >>> exact var. >>> >>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>> >>> Hi Guys, >>> >>> Here's my situation: >>> >>> The freeswitch server and my machine are behind the same LAN. If I >>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>> >>> However, external calls have no sound at all no matter whether this param >>> is commented out or not. >>> >>> Please kindly let me know what other params to set to resolve this issue. >>> >>> Thanks, >>> -Jingwei >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/95ffa9af/attachment-0002.html From jingwei.yang at gmail.com Wed Jun 24 22:31:35 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 13:31:35 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> Message-ID: <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> Hi Chris, thanks for your help. Here's my client.xml On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: > Please provide your client.xml detail with confidential information > crossout, I have gtalk client and server working properly behind the NAT. > I should be able to help you. > > Chris > > > On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: > >> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >> audio this time no matter what value disable-rtp-auto-adjust is... >> >> >> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >> >>> uncomment ext-rtp-ip >>> >>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>> >>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put >>> "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>> parameter has (true or false), local IP is able to hear the echo but >>> external ones still have no audio. >>> >>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>> >>>> search wiki from sth. like disable_rtp_autoajust , I don't remember the >>>> exact var. >>>> >>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>> >>>> Hi Guys, >>>> >>>> Here's my situation: >>>> >>>> The freeswitch server and my machine are behind the same LAN. If I >>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>> >>>> However, external calls have no sound at all no matter whether this >>>> param is commented out or not. >>>> >>>> Please kindly let me know what other params to set to resolve this >>>> issue. >>>> >>>> Thanks, >>>> -Jingwei >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a8de1d86/attachment-0002.html From msc at freeswitch.org Wed Jun 24 22:38:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Jun 2009 22:38:21 -0700 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> Message-ID: <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> FYI, Any language that can establish a network socket and send/receive information over that socket can be used to control FS. FreeSWITCH comes with ESL - the event socket library - that can abstract away some of the grunt work, but there isn't a Java one that I'm aware of. -MC On Wed, Jun 24, 2009 at 7:52 PM, Jingwei Yang wrote: > Hi Folks, > > I understand freeSwitch is supporting a couple of languages for call > controls like Lua, Javascript, Perl, Java... However, after digging into the > detailed wiki pages, I found out the codes written in those languages can > only be executed via the freeswitch console. I was wondering whether it's > possible to run FreeSwitch at backend and have a piece of Java program > (outside the freeSwitch console) to invoke freeSwitch commands? > > Thanks, > -Jingwei > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090624/c9b67a7b/attachment-0002.html From jingwei.yang at gmail.com Wed Jun 24 22:53:24 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 25 Jun 2009 13:53:24 +0800 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> Message-ID: <13529f9d0906242253sa25bf71k237b6e68afed8c04@mail.gmail.com> Thanks Michael, fs_eslib sounds the one for Java. I'll give it a try. On Thu, Jun 25, 2009 at 1:38 PM, Michael Collins wrote: > FYI, > Any language that can establish a network socket and send/receive > information over that socket can be used to control FS. FreeSWITCH comes > with ESL - the event socket library - that can abstract away some of the > grunt work, but there isn't a Java one that I'm aware of. > -MC > > On Wed, Jun 24, 2009 at 7:52 PM, Jingwei Yang wrote: > >> Hi Folks, >> >> I understand freeSwitch is supporting a couple of languages for call >> controls like Lua, Javascript, Perl, Java... However, after digging into the >> detailed wiki pages, I found out the codes written in those languages can >> only be executed via the freeswitch console. I was wondering whether it's >> possible to run FreeSwitch at backend and have a piece of Java program >> (outside the freeSwitch console) to invoke freeSwitch commands? >> >> Thanks, >> -Jingwei >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/dc5ed2e0/attachment-0002.html From peter.olsson at visionutveckling.se Thu Jun 25 00:36:47 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 25 Jun 2009 09:36:47 +0200 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d36d4319/attachment-0002.html From brian at freeswitch.org Thu Jun 25 01:15:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 03:15:39 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> Message-ID: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > I?ve been using FS as a gateway to our OCS server for some time. > It?s used just for testing, so it?s not really used every day. I > don?t know when, but after some trunk update (right now I running > r13945) of FS it doesn?t send the SIP traffic using tcp anymore (OCS > only accepts tcp or tls). > > My configuration is quite easy, I have a sofia gateway configured to > OCS, this has the parameter value="tcp"/> set in the config (nothing in the config has changed > for ages). Then in the dialplan I use this gateway to connect the > calls. When doing a siptrace I can see that the headers has > transport=tcp set correctly, but according to the trace it?s sent > using udp instead of tcp. > > Has something changed so I need to configure it in another way, or > is it just simply a bug? I just wanted to check this before issuing > a jira case and providing more specific information and debug traces > etc. > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/f366e706/attachment-0002.html From brian at freeswitch.org Thu Jun 25 01:20:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 03:20:04 -0500 Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking In-Reply-To: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> References: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> Message-ID: <85A736A7-AB9F-44D5-BB21-A20BB94B8A6A@freeswitch.org> check that s is nil. /b On Jun 24, 2009, at 8:12 PM, John Wehle wrote: > What's the recommended way to check if the session constructor was > successful (i.e. the number could be dialed)? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a3676691/attachment-0002.html From darklion11 at yahoo.com Thu Jun 25 01:39:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 01:39:38 -0700 (PDT) Subject: [Freeswitch-users] Originate works but dialplan does not work? In-Reply-To: <24196467.post@talk.nabble.com> References: <24181208.post@talk.nabble.com> <20090624094517.GA22279@jdc.jasonjgw.net> <24181933.post@talk.nabble.com> <2FCA1843-E7D7-4832-BC7C-94B4EEC1DC8A@gmail.com> <24196467.post@talk.nabble.com> Message-ID: <24199253.post@talk.nabble.com> Thanks a lot it works for me... Edmar Cruz wrote: > > If then, what bridge i shall call to? > > Like this? > > > > > dujinfang wrote: >> >> put your extension in dialplan/public.xml instead of sip_profiles/ >> external/myprofile.xml >> >> >> On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: >> >>> >>> Ooops.. Sorry wrong spelling... Same issue >>> >>> Jason White-14 wrote: >>>> >>>> Edmar Cruz wrote: >>>> >>>>> Here is my dialplan on sip_profiles/external/myprofile.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>> >>>> The above should be $1 not @1 >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24199253.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Thu Jun 25 03:16:38 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 03:16:38 -0700 (PDT) Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? Message-ID: <24200644.post@talk.nabble.com> Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? Thanks -- View this message in context: http://www.nabble.com/How-to-change-database-of-freeswitch-cdr-to-MySQL--tp24200644p24200644.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From chris.chen2004 at gmail.com Thu Jun 25 06:02:59 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 09:02:59 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> Message-ID: <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Two questions for you: 1) Do you have extension "888" in your public context? 2)Can you put your internal Ip address of FS in rtp-ip instead of $${bind_server_ip} just to make sure it get the right IP? 3) is not really required at least for my working setup behind the NAT router. On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang wrote: > Hi Chris, thanks for your help. Here's my client.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: > >> Please provide your client.xml detail with confidential information >> crossout, I have gtalk client and server working properly behind the NAT. >> I should be able to help you. >> >> Chris >> >> >> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: >> >>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >>> audio this time no matter what value disable-rtp-auto-adjust is... >>> >>> >>> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >>> >>>> uncomment ext-rtp-ip >>>> >>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>>> >>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put >>>> "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>>> parameter has (true or false), local IP is able to hear the echo but >>>> external ones still have no audio. >>>> >>>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>>> >>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember the >>>>> exact var. >>>>> >>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>>> >>>>> Hi Guys, >>>>> >>>>> Here's my situation: >>>>> >>>>> The freeswitch server and my machine are behind the same LAN. If I >>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>>> >>>>> However, external calls have no sound at all no matter whether this >>>>> param is commented out or not. >>>>> >>>>> Please kindly let me know what other params to set to resolve this >>>>> issue. >>>>> >>>>> Thanks, >>>>> -Jingwei >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/f3645c9e/attachment-0002.html From chris.chen2004 at gmail.com Thu Jun 25 06:07:50 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 09:07:50 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> Message-ID: <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ 103 at 192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: > Please open a jira and attach sip traces of register and phone calls. > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > > I?ve been using FS as a gateway to our OCS server for some time. It?s used > just for testing, so it?s not really used every day. I don?t know when, but > after some trunk update (right now I running r13945) of FS it doesn?t send > the SIP traffic using tcp anymore (OCS only accepts tcp or tls). > > My configuration is quite easy, I have a sofia gateway configured to OCS, > this has the parameter set in > the config (nothing in the config has changed for ages). Then in the > dialplan I use this gateway to connect the calls. When doing a siptrace I > can see that the headers has transport=tcp set correctly, but according to > the trace it?s sent using udp instead of tcp. > > Has something changed so I need to configure it in another way, or is it > just simply a bug? I just wanted to check this before issuing a jira case > and providing more specific information and debug traces etc. > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/7233dfd6/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 25 06:18:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 08:18:30 -0500 Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking In-Reply-To: <85A736A7-AB9F-44D5-BB21-A20BB94B8A6A@freeswitch.org> References: <200906250112.n5P1CHnX003383@jwlab.FEITH.COM> <85A736A7-AB9F-44D5-BB21-A20BB94B8A6A@freeswitch.org> Message-ID: <191c3a030906250618t145492a9x4e8de5d923dcc7ae@mail.gmail.com> and that s.ready() is true On Thu, Jun 25, 2009 at 3:20 AM, Brian West wrote: > check that s is nil. > /b > > On Jun 24, 2009, at 8:12 PM, John Wehle wrote: > > What's the recommended way to check if the session constructor was > successful (i.e. the number could be dialed)? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6aa1a36d/attachment-0002.html From brian at freeswitch.org Thu Jun 25 07:15:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 09:15:16 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> Message-ID: <1FA565E6-4BCC-4C02-93DB-CD3409119F41@freeswitch.org> Please open a jira and attach sip traces. /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > I am having the same issue, now the SIP trunk over TCP between FS > and Exchange 2007 UM just stops working, just stuck in a loop like > this: > > 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > > On my Exchange 2007 side nothing was changed which used to work fine > with FS > > Chris > > On Thu, Jun 25, 2009 at 4:15 AM, Brian West > wrote: > Please open a jira and attach sip traces of register and phone calls. > > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > >> I?ve been using FS as a gateway to our OCS server for some time. >> It?s used just for testing, so it?s not really used every day. I >> don?t know when, but after some trunk update (right now I running >> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >> (OCS only accepts tcp or tls). >> >> My configuration is quite easy, I have a sofia gateway configured >> to OCS, this has the parameter > value="tcp"/> set in the config (nothing in the config has changed >> for ages). Then in the dialplan I use this gateway to connect the >> calls. When doing a siptrace I can see that the headers has >> transport=tcp set correctly, but according to the trace it?s sent >> using udp instead of tcp. >> >> Has something changed so I need to configure it in another way, or >> is it just simply a bug? I just wanted to check this before issuing >> a jira case and providing more specific information and debug >> traces etc. >> >> /Peter >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/97a5716c/attachment-0002.html From dujinfang at gmail.com Thu Jun 25 08:16:33 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 25 Jun 2009 23:16:33 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Message-ID: > 3) is not > really required at least for my working setup behind the NAT router. > ok, this param is originally added for another problem http://jira.freeswitch.org/browse/MODENDP-198 . But I think it might be useful for this. From peter.olsson at visionutveckling.se Thu Jun 25 08:22:22 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 25 Jun 2009 17:22:22 +0200 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE4885C@cooper> Done, added as issue SFSIP-157. Regards, Peter Olsson Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a4333c332936913812693! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/a20735ee/attachment-0002.html From paul.degt at gmail.com Thu Jun 25 08:24:20 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 11:24:20 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <24200644.post@talk.nabble.com> References: <24200644.post@talk.nabble.com> Message-ID: <4A4396A4.2060701@gmail.com> You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: > Hi, > > > How can i change my freeswitch database instead of CSV file, I make it > mysql. Can you tell me how? > > > Thanks > From anthony.minessale at gmail.com Thu Jun 25 08:31:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 10:31:48 -0500 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Message-ID: <191c3a030906250831m15208063j6314a6be5f51d80@mail.gmail.com> if you are behind nat , you will not want to disable auto-adjust that is what the feature is there to help you with. On Thu, Jun 25, 2009 at 10:16 AM, Seven Du wrote: > > 3) is not > > really required at least for my working setup behind the NAT router. > > > > ok, this param is originally added for another problem > http://jira.freeswitch.org/browse/MODENDP-198 > . But I think it might be useful for this. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/b5154148/attachment-0002.html From brian at freeswitch.org Thu Jun 25 08:41:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 10:41:05 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE4885C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE4885C@cooper> Message-ID: <1B1FC434-6B4A-4A54-A547-1486A99AD53B@freeswitch.org> Fixed revision 13948. /b On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote: > Done, added as issue SFSIP-157. > > Regards, > > Peter Olsson > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] F?r Brian West > Skickat: den 25 juni 2009 10:16 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp > anymore... > > Please open a jira and attach sip traces of register and phone calls. > > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > > > I?ve been using FS as a gateway to our OCS server for some time. > It?s used just for testing, so it?s not really used every day. I > don?t know when, but after some trunk update (right now I running > r13945) of FS it doesn?t send the SIP traffic using tcp anymore (OCS > only accepts tcp or tls). > > My configuration is quite easy, I have a sofia gateway configured to > OCS, this has the parameter value="tcp"/> set in the config (nothing in the config has changed > for ages). Then in the dialplan I use this gateway to connect the > calls. When doing a siptrace I can see that the headers has > transport=tcp set correctly, but according to the trace it?s sent > using udp instead of tcp. > > Has something changed so I need to configure it in another way, or > is it just simply a bug? I just wanted to check this before issuing > a jira case and providing more specific information and debug traces > etc. > > /Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > !DSPAM:4a4333c332936913812693! > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/172b4300/attachment-0002.html From brian at freeswitch.org Thu Jun 25 08:49:40 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 10:49:40 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A4396A4.2060701@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> Message-ID: <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > You can use FS XML Curl - FS sends XML CDRs to a web server of your > choice, and there you do whatever you want with these CDRs, like store > in a database. > There are also pre-built solutions available, check here: > http://versafon.com/versafonweb/CommercialSupport.jsp > > Edmar Cruz wrote: >> Hi, >> >> >> How can i change my freeswitch database instead of CSV file, I >> make it >> mysql. Can you tell me how? >> >> >> Thanks From brian at freeswitch.org Thu Jun 25 09:06:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:06:59 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> Message-ID: <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > I am having the same issue, now the SIP trunk over TCP between FS > and Exchange 2007 UM just stops working, just stuck in a loop like > this: > > 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 > entering state [calling][0] > > On my Exchange 2007 side nothing was changed which used to work fine > with FS > > Chris > > On Thu, Jun 25, 2009 at 4:15 AM, Brian West > wrote: > Please open a jira and attach sip traces of register and phone calls. > > /b > > On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: > >> I?ve been using FS as a gateway to our OCS server for some time. >> It?s used just for testing, so it?s not really used every day. I >> don?t know when, but after some trunk update (right now I running >> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >> (OCS only accepts tcp or tls). >> >> My configuration is quite easy, I have a sofia gateway configured >> to OCS, this has the parameter > value="tcp"/> set in the config (nothing in the config has changed >> for ages). Then in the dialplan I use this gateway to connect the >> calls. When doing a siptrace I can see that the headers has >> transport=tcp set correctly, but according to the trace it?s sent >> using udp instead of tcp. >> >> Has something changed so I need to configure it in another way, or >> is it just simply a bug? I just wanted to check this before issuing >> a jira case and providing more specific information and debug >> traces etc. >> >> /Peter >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/e21e898a/attachment-0002.html From paul.degt at gmail.com Thu Jun 25 09:15:11 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 12:15:11 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> Message-ID: <4A43A28F.7020404@gmail.com> I am an employee, why? Brian West wrote: > Are you the owner of Versafon? > > /b > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of your >> choice, and there you do whatever you want with these CDRs, like store >> in a database. >> There are also pre-built solutions available, check here: >> http://versafon.com/versafonweb/CommercialSupport.jsp >> >> Edmar Cruz wrote: >> >>> Hi, >>> >>> >>> How can i change my freeswitch database instead of CSV file, I >>> make it >>> mysql. Can you tell me how? >>> >>> >>> Thanks >>> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chris.chen2004 at gmail.com Thu Jun 25 09:24:24 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 12:24:24 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> Message-ID: <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ 103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: > Can you verify that this is fixed... I think its related to the same > issue... > /b > > On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > > I am having the same issue, now the SIP trunk over TCP between FS and > Exchange 2007 UM just stops working, just stuck in a loop like this: > > 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/external/103 at 192.168.0.250 > 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ > 103 at 192.168.0.250 entering state [calling][0] > > On my Exchange 2007 side nothing was changed which used to work fine with > FS > > Chris > > On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: > >> Please open a jira and attach sip traces of register and phone calls. >> /b >> >> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >> >> I?ve been using FS as a gateway to our OCS server for some time. It?s >> used just for testing, so it?s not really used every day. I don?t know when, >> but after some trunk update (right now I running r13945) of FS it doesn?t >> send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). >> >> My configuration is quite easy, I have a sofia gateway configured to OCS, >> this has the parameter set in >> the config (nothing in the config has changed for ages). Then in the >> dialplan I use this gateway to connect the calls. When doing a siptrace I >> can see that the headers has transport=tcp set correctly, but according to >> the trace it?s sent using udp instead of tcp. >> >> Has something changed so I need to configure it in another way, or is it >> just simply a bug? I just wanted to check this before issuing a jira case >> and providing more specific information and debug traces etc. >> >> /Peter >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/9fe8a8a6/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 25 09:27:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 11:27:53 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43A28F.7020404@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> Message-ID: <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt wrote: > I am an employee, why? > > Brian West wrote: > > Are you the owner of Versafon? > > > > /b > > > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > > > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of your > >> choice, and there you do whatever you want with these CDRs, like store > >> in a database. > >> There are also pre-built solutions available, check here: > >> http://versafon.com/versafonweb/CommercialSupport.jsp > >> > >> Edmar Cruz wrote: > >> > >>> Hi, > >>> > >>> > >>> How can i change my freeswitch database instead of CSV file, I > >>> make it > >>> mysql. Can you tell me how? > >>> > >>> > >>> Thanks > >>> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/cabaf5cf/attachment-0002.html From peter.olsson at visionutveckling.se Thu Jun 25 09:28:47 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 25 Jun 2009 18:28:47 +0200 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E229@cooper> What can I say - you guys provide far much better (and quicker) support then any commersial solution :) Thanks for the help! /Peter ________________________________ Fr?n: Brian West Skickat: den 25 juni 2009 17:53 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Fixed revision 13948. /b On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote: Done, added as issue SFSIP-157. Regards, Peter Olsson Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I?ve been using FS as a gateway to our OCS server for some time. It?s used just for testing, so it?s not really used every day. I don?t know when, but after some trunk update (right now I running r13945) of FS it doesn?t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it?s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a439d8f32931361515932! From anthony.minessale at gmail.com Thu Jun 25 09:35:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 11:35:15 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> Message-ID: <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen wrote: > I upgraded to 13950, still the same, keeping the same loop like the console > log showing: > 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ > 103 at 192.168.0.250 entering state [calling][0] > 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > > Chris > > > On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: > >> Can you verify that this is fixed... I think its related to the same >> issue... >> /b >> >> On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: >> >> I am having the same issue, now the SIP trunk over TCP between FS and >> Exchange 2007 UM just stops working, just stuck in a loop like this: >> >> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ >> 103 at 192.168.0.250 entering state [calling][0] >> >> On my Exchange 2007 side nothing was changed which used to work fine with >> FS >> >> Chris >> >> On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: >> >>> Please open a jira and attach sip traces of register and phone calls. >>> /b >>> >>> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >>> >>> I?ve been using FS as a gateway to our OCS server for some time. It?s >>> used just for testing, so it?s not really used every day. I don?t know when, >>> but after some trunk update (right now I running r13945) of FS it doesn?t >>> send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). >>> >>> My configuration is quite easy, I have a sofia gateway configured to OCS, >>> this has the parameter set in >>> the config (nothing in the config has changed for ages). Then in the >>> dialplan I use this gateway to connect the calls. When doing a siptrace I >>> can see that the headers has transport=tcp set correctly, but according to >>> the trace it?s sent using udp instead of tcp. >>> >>> Has something changed so I need to configure it in another way, or is it >>> just simply a bug? I just wanted to check this before issuing a jira case >>> and providing more specific information and debug traces etc. >>> >>> /Peter >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d54749a6/attachment-0002.html From harmeet at litatel.com Thu Jun 25 09:35:49 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Thu, 25 Jun 2009 12:35:49 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: I didn't see the last message from Paul as advertizement. He was explaining about the FS XML Curl. I think he may have mentioned the link to versafon even if he didn't work there. On Thu, Jun 25, 2009 at 12:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > since you are advertising on our site regularly now perhaps you could ask > your boss to sponsor the project. > > > > On Thu, Jun 25, 2009 at 11:15 AM, paul.degt wrote: > >> I am an employee, why? >> >> Brian West wrote: >> > Are you the owner of Versafon? >> > >> > /b >> > >> > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: >> > >> > >> >> You can use FS XML Curl - FS sends XML CDRs to a web server of your >> >> choice, and there you do whatever you want with these CDRs, like store >> >> in a database. >> >> There are also pre-built solutions available, check here: >> >> http://versafon.com/versafonweb/CommercialSupport.jsp >> >> >> >> Edmar Cruz wrote: >> >> >> >>> Hi, >> >>> >> >>> >> >>> How can i change my freeswitch database instead of CSV file, I >> >>> make it >> >>> mysql. Can you tell me how? >> >>> >> >>> >> >>> Thanks >> >>> >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6fc0b914/attachment-0002.html From brian at freeswitch.org Thu Jun 25 09:35:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:35:22 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> Message-ID: <8B617782-8E40-42F4-B132-6C1CA5B98137@freeswitch.org> I see what is wrong you're running it all on the same IP and the 302 is back to the same IP but a different port I need to fix that logic to compare the port number also. /b On Jun 25, 2009, at 11:24 AM, Chris Chen wrote: > I upgraded to 13950, still the same, keeping the same loop like the > console log showing: > 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > > Chris > > On Thu, Jun 25, 2009 at 12:06 PM, Brian West > wrote: > Can you verify that this is fixed... I think its related to the same > issue... > > /b > > On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > >> I am having the same issue, now the SIP trunk over TCP between FS >> and Exchange 2007 UM just stops working, just stuck in a loop like >> this: >> >> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> >> On my Exchange 2007 side nothing was changed which used to work >> fine with FS >> >> Chris >> >> On Thu, Jun 25, 2009 at 4:15 AM, Brian West >> wrote: >> Please open a jira and attach sip traces of register and phone calls. >> >> /b >> >> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >> >>> I?ve been using FS as a gateway to our OCS server for some time. >>> It?s used just for testing, so it?s not really used every day. I >>> don?t know when, but after some trunk update (right now I running >>> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >>> (OCS only accepts tcp or tls). >>> >>> My configuration is quite easy, I have a sofia gateway configured >>> to OCS, this has the parameter >> value="tcp"/> set in the config (nothing in the config has changed >>> for ages). Then in the dialplan I use this gateway to connect the >>> calls. When doing a siptrace I can see that the headers has >>> transport=tcp set correctly, but according to the trace it?s sent >>> using udp instead of tcp. >>> >>> Has something changed so I need to configure it in another way, or >>> is it just simply a bug? I just wanted to check this before >>> issuing a jira case and providing more specific information and >>> debug traces etc. >>> >>> /Peter >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/89d10c10/attachment-0002.html From msc at freeswitch.org Thu Jun 25 09:45:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 25 Jun 2009 09:45:25 -0700 Subject: [Freeswitch-users] FreeSwitch at backend In-Reply-To: <13529f9d0906242253sa25bf71k237b6e68afed8c04@mail.gmail.com> References: <13529f9d0906241952yfd97215ya09d52b0c61d2d71@mail.gmail.com> <87f2f3b90906242238x17b31c16pbef21f53df29e72f@mail.gmail.com> <13529f9d0906242253sa25bf71k237b6e68afed8c04@mail.gmail.com> Message-ID: <87f2f3b90906250945o2fa84e03t9f92fbede192dce2@mail.gmail.com> On Wed, Jun 24, 2009 at 10:53 PM, Jingwei Yang wrote: > Thanks Michael, fs_eslib sounds the one for Java. I'll give it a try. > Let us know how it goes. We like success stories! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/9bbf4d49/attachment-0002.html From edpimentl at gmail.com Thu Jun 25 09:48:32 2009 From: edpimentl at gmail.com (EdPimentl) Date: Thu, 25 Jun 2009 12:48:32 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitch Sincerely, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/57290535/attachment-0002.html From brian at freeswitch.org Thu Jun 25 09:52:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:52:09 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> Message-ID: <0E44A2AB-8232-401F-90FA-B0A87E3568FE@freeswitch.org> And possibly present the word FreeSWITCH in the proper case! ;) /b On Jun 25, 2009, at 11:48 AM, EdPimentl wrote: > To be fair ... > when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp > a common courtesy would be to provide a link to Freeswitch > > Sincerely, > -E Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d07b3692/attachment-0002.html From brian at freeswitch.org Thu Jun 25 09:53:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 11:53:02 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> Message-ID: I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: > are you redirecting it to yourself by any chance because of some > proxy in your network? > > > On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen > wrote: > I upgraded to 13950, still the same, keeping the same loop like the > console log showing: > 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 > entering state [calling][0] > 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp > Setting proxy route to sofia/internal/103 at 192.168.0.250 > > Chris > > > On Thu, Jun 25, 2009 at 12:06 PM, Brian West > wrote: > Can you verify that this is fixed... I think its related to the same > issue... > > /b > > On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: > >> I am having the same issue, now the SIP trunk over TCP between FS >> and Exchange 2007 UM just stops working, just stuck in a loop like >> this: >> >> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/external/103 at 192.168.0.250 >> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 >> entering state [calling][0] >> >> On my Exchange 2007 side nothing was changed which used to work >> fine with FS >> >> Chris >> >> On Thu, Jun 25, 2009 at 4:15 AM, Brian West >> wrote: >> Please open a jira and attach sip traces of register and phone calls. >> >> /b >> >> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >> >>> I?ve been using FS as a gateway to our OCS server for some time. >>> It?s used just for testing, so it?s not really used every day. I >>> don?t know when, but after some trunk update (right now I running >>> r13945) of FS it doesn?t send the SIP traffic using tcp anymore >>> (OCS only accepts tcp or tls). >>> >>> My configuration is quite easy, I have a sofia gateway configured >>> to OCS, this has the parameter >> value="tcp"/> set in the config (nothing in the config has changed >>> for ages). Then in the dialplan I use this gateway to connect the >>> calls. When doing a siptrace I can see that the headers has >>> transport=tcp set correctly, but according to the trace it?s sent >>> using udp instead of tcp. >>> >>> Has something changed so I need to configure it in another way, or >>> is it just simply a bug? I just wanted to check this before >>> issuing a jira case and providing more specific information and >>> debug traces etc. >>> >>> /Peter >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/bd7e64df/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 25 09:54:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 11:54:07 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: <191c3a030906250954j671a92d4p50ccdc83dad89485@mail.gmail.com> It's a suggestion not a demand. If you follow the link it's a list of products and services related to FS. We prefer that anyone who wants to sell stuff made from FS or as an accessory to FS would consider sponsoring the project or ClueCon. We provide both commercial and free support for FreeSWITCH and the amount of free help we have time to give is directly impacted by how many people who use FreeSWITCH for commercial purposes give back to us in the form of volunteer developers, support contracts and sponsorship. Trust me, if we don't ask very few will realize it on their own. On Thu, Jun 25, 2009 at 11:35 AM, Harmeet Singh wrote: > I didn't see the last message from Paul as advertizement. He was explaining > about the FS XML Curl. I think he may have mentioned the link to versafon > even if he didn't work there. > > > On Thu, Jun 25, 2009 at 12:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> since you are advertising on our site regularly now perhaps you could ask >> your boss to sponsor the project. >> >> >> >> On Thu, Jun 25, 2009 at 11:15 AM, paul.degt wrote: >> >>> I am an employee, why? >>> >>> Brian West wrote: >>> > Are you the owner of Versafon? >>> > >>> > /b >>> > >>> > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: >>> > >>> > >>> >> You can use FS XML Curl - FS sends XML CDRs to a web server of your >>> >> choice, and there you do whatever you want with these CDRs, like store >>> >> in a database. >>> >> There are also pre-built solutions available, check here: >>> >> http://versafon.com/versafonweb/CommercialSupport.jsp >>> >> >>> >> Edmar Cruz wrote: >>> >> >>> >>> Hi, >>> >>> >>> >>> >>> >>> How can i change my freeswitch database instead of CSV file, I >>> >>> make it >>> >>> mysql. Can you tell me how? >>> >>> >>> >>> >>> >>> Thanks >>> >>> >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6afeacd0/attachment-0002.html From msc at freeswitch.org Thu Jun 25 09:54:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 25 Jun 2009 09:54:35 -0700 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> Message-ID: <87f2f3b90906250954o7ef9da6dp48fae6a140fc97aa@mail.gmail.com> On Thu, Jun 25, 2009 at 9:48 AM, EdPimentl wrote: > To be fair ... > when mentioning Freeswitch here > http://versafon.com/versafonweb/Software.jsp > a common courtesy would be to provide a link to Freeswitch > Not only that but spelling "FreeSWITCH" correctly would be a nice touch, no? :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/98b87a47/attachment-0002.html From paul.degt at gmail.com Thu Jun 25 09:57:29 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 12:57:29 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> Message-ID: <4A43AC79.6080205@gmail.com> I talked about it but with the current state of VOIP market it's seems problematic. :-( I did not see my posts as strictly ads since we offer free software as well and also share my limited knowledge about FS. Anthony Minessale wrote: > since you are advertising on our site regularly now perhaps you could > ask your boss to sponsor the project. > > > On Thu, Jun 25, 2009 at 11:15 AM, paul.degt > wrote: > > I am an employee, why? > > Brian West wrote: > > Are you the owner of Versafon? > > > > /b > > > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > > > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of > your > >> choice, and there you do whatever you want with these CDRs, > like store > >> in a database. > >> There are also pre-built solutions available, check here: > >> http://versafon.com/versafonweb/CommercialSupport.jsp > >> > >> Edmar Cruz wrote: > >> > >>> Hi, > >>> > >>> > >>> How can i change my freeswitch database instead of CSV file, I > >>> make it > >>> mysql. Can you tell me how? > >>> > >>> > >>> Thanks > >>> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From paul.degt at gmail.com Thu Jun 25 10:02:37 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 13:02:37 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <0E44A2AB-8232-401F-90FA-B0A87E3568FE@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <0E44A2AB-8232-401F-90FA-B0A87E3568FE@freeswitch.org> Message-ID: <4A43ADAD.4040504@gmail.com> Done. :-) Brian West wrote: > And possibly present the word FreeSWITCH in the proper case! ;) > > /b > > On Jun 25, 2009, at 11:48 AM, EdPimentl wrote: > >> To be fair ... >> when mentioning Freeswitch >> here http://versafon.com/versafonweb/Software.jsp >> a common courtesy would be to provide a link to Freeswitch >> >> >> Sincerely, >> -E > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From paul.degt at gmail.com Thu Jun 25 10:04:18 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 13:04:18 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> Message-ID: <4A43AE12.70500@gmail.com> I will have to ask my boss about that, most probably he will ask same in return. EdPimentl wrote: > To be fair ... > when mentioning Freeswitch here > http://versafon.com/versafonweb/Software.jsp > a common courtesy would be to provide a link to Freeswitch > > > Sincerely, > -E > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Thu Jun 25 10:05:12 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 13:05:12 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> Message-ID: <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is working. Thanks for your great work. Chris On Thu, Jun 25, 2009 at 12:53 PM, Brian West wrote: > I found the problem... the fs_path refactor regression number 2 was just > fixed.. It was assuming the route_uri was the contact and making it stick to > the wrong place to send the invite... you should be able to update now and > it work correctly. > /b > > On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: > > are you redirecting it to yourself by any chance because of some proxy in > your network? > > > On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen wrote: > >> I upgraded to 13950, still the same, keeping the same loop like the >> console log showing: >> 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250;transport=tcp >> Setting proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ >> 103 at 192.168.0.250 entering state [calling][0] >> 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 >> sip:103 at 192.168.0.250 ;transport=tcp Setting >> proxy route to sofia/internal/103 at 192.168.0.250 >> >> Chris >> >> >> On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: >> >>> Can you verify that this is fixed... I think its related to the same >>> issue... >>> /b >>> >>> On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: >>> >>> I am having the same issue, now the SIP trunk over TCP between FS and >>> Exchange 2007 UM just stops working, just stuck in a loop like this: >>> >>> 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 >>> sip:103 at 192.168.0.250 ;transport=tcp Setting >>> proxy route to sofia/external/103 at 192.168.0.250 >>> 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ >>> 103 at 192.168.0.250 entering state [calling][0] >>> >>> On my Exchange 2007 side nothing was changed which used to work fine with >>> FS >>> >>> Chris >>> >>> On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: >>> >>>> Please open a jira and attach sip traces of register and phone calls. >>>> /b >>>> >>>> On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: >>>> >>>> I?ve been using FS as a gateway to our OCS server for some time. It?s >>>> used just for testing, so it?s not really used every day. I don?t know when, >>>> but after some trunk update (right now I running r13945) of FS it doesn?t >>>> send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). >>>> >>>> My configuration is quite easy, I have a sofia gateway configured to >>>> OCS, this has the parameter >>>> set in the config (nothing in the config has changed for ages). Then in the >>>> dialplan I use this gateway to connect the calls. When doing a siptrace I >>>> can see that the headers has transport=tcp set correctly, but according to >>>> the trace it?s sent using udp instead of tcp. >>>> >>>> Has something changed so I need to configure it in another way, or is it >>>> just simply a bug? I just wanted to check this before issuing a jira case >>>> and providing more specific information and debug traces etc. >>>> >>>> /Peter >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/96db9043/attachment-0002.html From intralanman at freeswitch.org Thu Jun 25 10:08:30 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 25 Jun 2009 13:08:30 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AE12.70500@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com> Message-ID: <4A43AF0E.6010809@freeswitch.org> On 06/25/2009 01:04 PM, paul.degt wrote: > I will have to ask my boss about that, most probably he will ask same in > That doesn't really make sense... FreeSWITCH isn't using or benefitting from your software... but yours is from FreeSWITCH -Ray From anthony.minessale at gmail.com Thu Jun 25 10:09:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 12:09:08 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AC79.6080205@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <4A43AC79.6080205@gmail.com> Message-ID: <191c3a030906251009n39c9dcb7j1b837a4ce84ca858@mail.gmail.com> Right which is why i said it was a "suggestion" not a demand. If you want to help answer email in here or answer questions on a regular basis that's just as valuable. if your boss has no money to sponsor maybe he can donate you =D On Thu, Jun 25, 2009 at 11:57 AM, paul.degt wrote: > I talked about it but with the current state of VOIP market it's seems > problematic. :-( > I did not see my posts as strictly ads since we offer free software as > well and also share my limited knowledge about FS. > > Anthony Minessale wrote: > > since you are advertising on our site regularly now perhaps you could > > ask your boss to sponsor the project. > > > > > > On Thu, Jun 25, 2009 at 11:15 AM, paul.degt > > wrote: > > > > I am an employee, why? > > > > Brian West wrote: > > > Are you the owner of Versafon? > > > > > > /b > > > > > > On Jun 25, 2009, at 10:24 AM, paul.degt wrote: > > > > > > > > >> You can use FS XML Curl - FS sends XML CDRs to a web server of > > your > > >> choice, and there you do whatever you want with these CDRs, > > like store > > >> in a database. > > >> There are also pre-built solutions available, check here: > > >> http://versafon.com/versafonweb/CommercialSupport.jsp > > >> > > >> Edmar Cruz wrote: > > >> > > >>> Hi, > > >>> > > >>> > > >>> How can i change my freeswitch database instead of CSV file, I > > >>> make it > > >>> mysql. Can you tell me how? > > >>> > > >>> > > >>> Thanks > > >>> > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/12d44479/attachment-0002.html From anthony.minessale at gmail.com Thu Jun 25 10:18:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 12:18:02 -0500 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AF0E.6010809@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com> <4A43AF0E.6010809@freeswitch.org> Message-ID: <191c3a030906251018y2727ad3cgf04a89049334a75b@mail.gmail.com> FreeSWITCH Solutions will soon be offering a product gallery where companies who use FS can become certified partners and display their products. On Thu, Jun 25, 2009 at 12:08 PM, Raymond Chandler < intralanman at freeswitch.org> wrote: > On 06/25/2009 01:04 PM, paul.degt wrote: > > I will have to ask my boss about that, most probably he will ask same in > > > > That doesn't really make sense... FreeSWITCH isn't using or benefitting > from your software... but yours is from FreeSWITCH > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/cd917142/attachment-0002.html From paul.degt at gmail.com Thu Jun 25 10:23:35 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 25 Jun 2009 13:23:35 -0400 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43AF0E.6010809@freeswitch.org> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com> <4A43AF0E.6010809@freeswitch.org> Message-ID: <4A43B297.7030005@gmail.com> FS users may well be benefiting - thus FS itself benefits as well indirectly. Raymond Chandler wrote: > On 06/25/2009 01:04 PM, paul.degt wrote: > >> I will have to ask my boss about that, most probably he will ask same in >> >> > > That doesn't really make sense... FreeSWITCH isn't using or benefitting > from your software... but yours is from FreeSWITCH > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From d at d-man.org Thu Jun 25 10:34:18 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 25 Jun 2009 10:34:18 -0700 Subject: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? In-Reply-To: <4A43B297.7030005@gmail.com> References: <24200644.post@talk.nabble.com> <4A4396A4.2060701@gmail.com> <3DEED6EA-0035-4B1C-8766-D8A973220B05@freeswitch.org> <4A43A28F.7020404@gmail.com> <191c3a030906250927s6fa1a497uc722ef89b998d701@mail.gmail.com> <9dc4a1670906250948v4e083c93iacc7aa105f5a6986@mail.gmail.com> <4A43AE12.70500@gmail.com><4A43AF0E.6010809@freeswitch.org> <4A43B297.7030005@gmail.com> Message-ID: <70AABAE3FDEC46B1953D0DAA17F7DE2E@test> Is it possible to unsusbcribe from specific threads on this list? Specifically I am looking for C code that removes useless banter so my brain doesn't hurt so much... -----Original Message----- From: paul.degt [mailto:paul.degt at gmail.com] Sent: Thursday, June 25, 2009 10:24 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? FS users may well be benefiting - thus FS itself benefits as well indirectly. Raymond Chandler wrote: > On 06/25/2009 01:04 PM, paul.degt wrote: > >> I will have to ask my boss about that, most probably he will ask same >> in >> >> > > That doesn't really make sense... FreeSWITCH isn't using or > benefitting from your software... but yours is from FreeSWITCH > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From d at d-man.org Thu Jun 25 10:35:29 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 25 Jun 2009 10:35:29 -0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> Message-ID: Did this work? Would love an update on this error/issue. _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, June 24, 2009 8:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway try adding before the bridge and report back results. Mike On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: Dear All, Look like nibblebill does't work with multiple gatreway. I try > before the bridge and report back results. > > Mike > > On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: > > Dear All, > > Look like nibblebill does't work with multiple gatreway. > I try > data="nibble_account=0838833133"/> > > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3] > sofia/external/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5] > sofia/external/6626734000 at 202.xxx.xxx.xxx> > > nibblebill not found nibble_rate > > But > > data="nibble_account=0838833133"/> > > session.execute("set", "import=nibble_account"); > session.execute("bridge", "{absolute_codec_string='GSM,G729'} > [nibble_rate=0.5,nibble_account=0838833133]sofia/external/xxxx at xxxx.xxx.xxx.xx > "); > > when call connected nibble do nothing i found heartbeat > > mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! > when call disconnect nibble update amont. > mod_nibblebill.c:478 Billing 16 secs > > I think nibble still not found variable channel. > > Let's me share more information > > I want to use nibblebill for callingcard. (i have develop billing by > myself). i plan to use javascript connect to ODBC > when customer call my script query balance and say. > and then i loop for get destination (my customer want to dial many > number). when i got number my script query > gateway from DB. i have 3 route and order by cost. > First plan i use > session.execute("bridge", > "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/ > xxxx at provder1| > [nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/ > xxxx at provder2"); > i modify nibblebill for match provider with my billing. > this case still fail. > > now i try > > if (session.ready()){ > s = new Session("{absolute_codec_string='GSM,G729'}sofia/ > external/xxx at provider1" > } > if (s.ready()){ > session.execute("set", "nibble_rate=2.5"); > session.execute("set", "nibble_account="+acaller); > session.execute("set", "hangup_after_bridge=false"); > session.execute("set", "provider_id="+dialprovider_id[1]); > bridge(session,s); > } > > and check hangup cause before try other provider. > > > > Please guide me it's right way or not ? > > > Dome C. > > > 2009/6/26 Darren Schreiber > Did this work? Would love an update on this error/issue. > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Wednesday, June 24, 2009 8:15 AM > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway > > try adding > > before the bridge and report back results. > > Mike > > On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: > >> Dear All, >> >> Look like nibblebill does't work with multiple gatreway. >> I try >> >> > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx >> |[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> nibblebill not found nibble_rate >> >> But >> >> >> > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx >> |sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> Work fine >> >> What's difference from set application and [] ? >> >> Best Regards. >> Dome C. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/55a9003a/attachment-0002.html From john at feith.com Thu Jun 25 12:35:55 2009 From: john at feith.com (John Wehle) Date: Thu, 25 Jun 2009 15:35:55 -0400 (EDT) Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking Message-ID: <200906251935.n5PJZtUt004125@jwlab.FEITH.COM> >> What's the recommended way to check if the session constructor was >> successful (i.e. the number could be dialed)? > check that s is nil. Doesn't work ... s is never nil. Type shows it as userdata even if Session failed. Specifically my test was: local s = freeswitch.Session ( "{ignore_early_media=true,origination_caller_id_name=" .. caller .. "}loopback/" .. destination .. "/default/XML") stream:write (type(s)) if s == nil then stream:write ("-ERR call failed\n") return end and I dialed an unreachable number. > and that s.ready() is true Checking s.ready() results in: [ERR] freeswitch_lua.cpp:102 session is not initalized if Session failed. What I'm looking for is a way to try to originate a call which doesn't throw ERR messages if the attempt fails. Explicitly calling session.originate seems to allow you to check if the call was successful ... is there a particular reason it's discouraged? I'm happy to avoid it if a better approach is available, however I'm having trouble finding one. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From anthony.minessale at gmail.com Thu Jun 25 12:50:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 25 Jun 2009 14:50:30 -0500 Subject: [Freeswitch-users] Originating a call from lua with rudimentary error checking In-Reply-To: <200906251935.n5PJZtUt004125@jwlab.FEITH.COM> References: <200906251935.n5PJZtUt004125@jwlab.FEITH.COM> Message-ID: <191c3a030906251250k4050dba9sdef0a3c633898cb5@mail.gmail.com> I think this is an oversight update to trunk and session.ready() should work as expected. On Thu, Jun 25, 2009 at 2:35 PM, John Wehle wrote: > >> What's the recommended way to check if the session constructor was > >> successful (i.e. the number could be dialed)? > > > check that s is nil. > > Doesn't work ... s is never nil. Type shows it as userdata > even if Session failed. Specifically my test was: > > local s = freeswitch.Session ( > "{ignore_early_media=true,origination_caller_id_name=" .. > caller .. "}loopback/" .. destination .. "/default/XML") > > stream:write (type(s)) > > if s == nil then > stream:write ("-ERR call failed\n") > return > end > > and I dialed an unreachable number. > > > and that s.ready() is true > > Checking s.ready() results in: > > [ERR] freeswitch_lua.cpp:102 session is not initalized > > if Session failed. > > What I'm looking for is a way to try to originate a call which doesn't > throw ERR messages if the attempt fails. > > Explicitly calling session.originate seems to allow you to check if > the call was successful ... is there a particular reason it's discouraged? > > I'm happy to avoid it if a better approach is available, however I'm > having trouble finding one. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/3eeea664/attachment-0002.html From vince.freeswitch at hightek.org Thu Jun 25 13:44:56 2009 From: vince.freeswitch at hightek.org (Vincent) Date: Thu, 25 Jun 2009 15:44:56 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> Message-ID: <20090625204456.GA45220@quark.hightek.org> On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > > > On Jun 23, 2009, at 10:15 PM, Vincent Stemen wrote: > > > Here is what I did and the results: > > > > ==================================================== > > Checked out the current trunk with svn. > > > > Patched /usr/include/sys/resource.h > > > > Since Dragonfly has fixed or will be fixing this future releases I > > patched the > > system header to add RLIMIT_AS rather than patching freeswitch to use > > RLIMIT_VMEM. > > Can we make a patch ifdefing on RLIMIT_AS to make this always work > without patches to system header files? Thanks for the responses Michael. I did this for attempting to compile freeswitch-1.0.3 and trunk as of a couple months ago. It would not apply to the current freeswitch trunk though. Apparently there have been changes to that area of the code. Since RLIMIT_AS is apparently a posix standard definition, I think this is fixed in Dragonfly HEAD and should not be a problem with future releases. I could go ahead and make a new patch when I get a chance if you still want me to, for compatibility with older Dragonfly releases. > > Compiling > > ========= > > > > Still lots of warnings of: > > warning: return makes pointer from integer without a cast > > > > Errors: > > It is apparently not checking return codes from make. It continues > > even when > > there are errors. Is this intentional?? > > > > su_alloc.c: In function `su_salloc': > > su_alloc.c:1518: warning: return makes pointer from integer without > > a cast > > gmake[9]: *** [su_alloc.lo] Error 1 > > gmake[8]: *** [all] Error 2 > > Making all in features > > LTCOMPILE features.lo > > ... > > > > Making all in sresolv > > LTCOMPILE sres.lo > > LTCOMPILE sres_cache.lo > > LTCOMPILE sres_blocking.lo > > LTCOMPILE sresolv.lo > > LTCOMPILE sres_sip.lo > > sres_sip.c: In function `sres_sip_new': > > sres_sip.c:267: warning: return makes pointer from integer without > > a cast > > gmake[8]: *** [sres_sip.lo] Error 1 > > Making all in ipt > > LTCOMPILE base64.lo > > LTCOMPILE token64.lo > > LINK libipt.la > > ... > > > > There are about 12 errors of this nature before ending with > > > > Making all in nua > > LTCOMPILE nua.lo > > nua.c: In function `nua_create': > > nua.c:141: warning: return makes pointer from integer without a cast > > nua.c:144: warning: return makes pointer from integer without a cast > > gmake[9]: *** [nua.lo] Error 1 > > gmake[8]: *** [all] Error 2 > > gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed > > by `libsofia-sip-ua.la'. Stop. > > gmake[7]: *** [all-recursive] Error 1 > > Making all in packages > > gmake[6]: *** [all-recursive] Error 1 > > gmake[5]: *** [all] Error 2 > > gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > > freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- > > ua.la] Error 2 > > gmake[3]: *** [mod_sofia-all] Error 1 > > gmake[2]: *** [all-recursive] Error 1 > > Making all in build > > +-------- FreeSWITCH Build Complete -----------+ > > + FreeSWITCH has been successfully built. + > > + Install by running: + > > + + > > + gmake install + > > +----------------------------------------------+ > > gmake[1]: *** [all-recursive] Error 1 > > gmake: *** [all] Error 2 > > > > Can you post a bug to Jira.freeswitch.org with all these warnings, > even better with patches to fix it. > > > > > It says it has been successfully built. Apparently part of the same > > problem of > > not checking the return codes. > > > > Patches to fix this appreciated Heh :-) OK. If I get it working and we end up using freeswitch, I will probably take a look at seeing if I can fix some or all of these warnings and create patches. > > It does not say what most of the errors are except for near the last > > when it > > says > > No rule to make target `iptsec/libiptsec.la' > > > > It just says "Error 1" or Error 2" which does not tell me what the > > problem is. From vince.freeswitch at hightek.org Thu Jun 25 14:49:10 2009 From: vince.freeswitch at hightek.org (Vincent) Date: Thu, 25 Jun 2009 16:49:10 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090624031950.GD2623@hijacked.us> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <20090624031950.GD2623@hijacked.us> Message-ID: <20090625214910.GB45220@quark.hightek.org> On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote: > On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: > > Ok. I did this. > > > > Compilation still failed but there are significant improvements since > > the last time. > > > > Here is what I did and the results: > > > > It looks like some the games that sofia plays with errno makes Dragonfly > unhappy. I also noticed that where the code checks for BSD-like systems > (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is > omitted, so obviously one of the first steps would be to fix that (if > applicable). > > If you disable mod_sofia in modules conf, do the rest of the default > modules build OK? OK. I commented out endpoints/mod_sofia. It looks like that eliminated all the errors except the one I get at the end. making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/nsprpub/dist/include/nspr/.: File exists gmake[9]: *** [export] Error 1 gmake[8]: *** [export] Error 2 gmake[7]: *** [export] Error 2 gmake[6]: *** [export] Error 2 gmake[5]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 So, it looks like most all the problems, except for that symlink error, including the pointer cast warnings, are related to the sofia module. I notice a lot of the modules seem to be redirecting the output somewhere. Not only do they just say Error 1 or Error 2 when there is an error, they also do not show the compile commands. They just output something like "Making built-sources in su" or "Compiling src/switch_apr.c ...". Is there a log file somewhere that contains the actual compile commands and error output so you can find out what happened when there is a error? Or perhaps a configuration to enable it to come out on the console? > For the record, DragonFly and FreeBSD have rather seriously diverged at > this point, DragonFly forked from FreeBSD back in the 4.10 days or so > and has changed a *lot* of things since, so I don't think it's gonna be > quite as easy as you expected (but it's far from impossible either). > > Andrew True, architecturally Dragonfly is becoming very different. They seem to be trying to maintain fairly good API compatibility though. Enough to constantly allow them to bring across major sub-systems, such as sound and SATA drivers, etc, from FreeBSD. So far, they have been pretty good about correcting it as soon as possible whenever one of us finds an incompatibility (Such as the RLIMIT_AS issue). Usually, all I have to do is add "-D__FreeBSD__" to CFLAGS and CPPFLAGS to compile packages that do not natively know about Dragonfly yet. Which is what I am doing with freeswitch. From drago at windstream.net Thu Jun 25 14:55:28 2009 From: drago at windstream.net (Drago Totev) Date: Thu, 25 Jun 2009 17:55:28 -0400 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> Message-ID: <000001c9f5df$a7823410$f6869c30$@net> Will it be a Windows build with the fix available soon? Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Chen Sent: Thursday, June 25, 2009 1:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is working. Thanks for your great work. Chris On Thu, Jun 25, 2009 at 12:53 PM, Brian West wrote: I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen wrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/103 at 192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:103 at 192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/103 at 192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/103 at 192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/637e70f7/attachment-0002.html From brian at freeswitch.org Thu Jun 25 14:57:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 16:57:48 -0500 Subject: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... In-Reply-To: <000001c9f5df$a7823410$f6869c30$@net> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEE487ED@cooper> <507898380906250607y11bacc70u5e9079b4d6a7ed3b@mail.gmail.com> <48024953-DFF4-4984-8F16-B15AF80CB14C@freeswitch.org> <507898380906250924k3ea452b0s7a6c4930d0045821@mail.gmail.com> <191c3a030906250935m13da5edfued50b65ceda80063@mail.gmail.com> <507898380906251005sa6e5d46ua5c2c5500cc7b8ef@mail.gmail.com> <000001c9f5df$a7823410$f6869c30$@net> Message-ID: I don't think the windows build was updated to include the bug... but you can build it with MSVC Express Edition which is Free from Microsoft. /b On Jun 25, 2009, at 4:55 PM, Drago Totev wrote: > Will it be a Windows build with the fix available soon? > > Drago Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/08834a2f/attachment-0002.html From mike at jerris.com Thu Jun 25 15:06:04 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 25 Jun 2009 18:06:04 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090625214910.GB45220@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <20090624031950.GD2623@hijacked.us> <20090625214910.GB45220@quark.hightek.org> Message-ID: <9E2119E8-99A3-40B9-960F-539B881CF1EE@jerris.com> On Jun 25, 2009, at 5:49 PM, Vincent wrote: > On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote: >> On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: >>> Ok. I did this. >>> >>> Compilation still failed but there are significant improvements >>> since >>> the last time. >>> >>> Here is what I did and the results: >>> >> >> It looks like some the games that sofia plays with errno makes >> Dragonfly >> unhappy. I also noticed that where the code checks for BSD-like >> systems >> (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, >> DragonFly is >> omitted, so obviously one of the first steps would be to fix that (if >> applicable). >> >> If you disable mod_sofia in modules conf, do the rest of the default >> modules build OK? > > OK. I commented out endpoints/mod_sofia. It looks like that > eliminated > all the errors except the one I get at the end. > > making all mod_spidermonkey > cd config; gmake -j1 export > cd pr; gmake -j1 export > cd include; gmake export > cd md; gmake export > ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ > ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/nsprpub/ > dist/include/nspr/.: File exists > gmake[9]: *** [export] Error 1 > gmake[8]: *** [export] Error 2 > gmake[7]: *** [export] Error 2 > gmake[6]: *** [export] Error 2 > gmake[5]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > freeswitch-20090623/libs/js/libjs.la] Error 2 > gmake[4]: *** [all] Error 1 > gmake[3]: *** [mod_spidermonkey-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 you can also comment out that module and see if you get further. > > > So, it looks like most all the problems, except for that symlink > error, > including the pointer cast warnings, are related to the sofia module. > > I notice a lot of the modules seem to be redirecting the output > somewhere. > > Not only do they just say Error 1 or Error 2 when there is an error, > they > also do not show the compile commands. They just output something > like > "Making built-sources in su" or "Compiling src/switch_apr.c ...". Is > there a log file somewhere that contains the actual compile commands > and > error output so you can find out what happened when there is a error? > Or perhaps a configuration to enable it to come out on the console? VERBOSE=1 gmake >> For the record, DragonFly and FreeBSD have rather seriously >> diverged at >> this point, DragonFly forked from FreeBSD back in the 4.10 days or so >> and has changed a *lot* of things since, so I don't think it's >> gonna be >> quite as easy as you expected (but it's far from impossible either). >> >> Andrew > > True, architecturally Dragonfly is becoming very different. They seem > to be trying to maintain fairly good API compatibility though. Enough > to constantly allow them to bring across major sub-systems, such as > sound and SATA drivers, etc, from FreeBSD. So far, they have been > pretty good about correcting it as soon as possible whenever one of us > finds an incompatibility (Such as the RLIMIT_AS issue). > > Usually, all I have to do is add "-D__FreeBSD__" to CFLAGS and > CPPFLAGS > to compile packages that do not natively know about Dragonfly yet. > Which is what I am doing with freeswitch. > From Richard.Lamkin at mettoni.com Thu Jun 25 15:08:29 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 25 Jun 2009 23:08:29 +0100 Subject: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF46138053C7833@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF46138053C7CC9@nickel.mettonigroup.com> Dear Anthony and Brian, Firstly please accept my apologies for wasting your time. Brian's request for the SVN number prompted me to realise I was running with an out of date version of FS. When I synced up to the head of the trunk and reran my tests the scenario I described below worked perfectly with no stuck calls. Therefore the sequence Park, Ringing (ring back), Redirect using the event API has provided me with the automated redirection I was seeking. Thank you for your advice earlier this week and prompt turnaround of fixes for the problems I encountered with bridged and deflected calls. Regards Richard Lamkin Richard.lamkin at mettoni.com From: Brian West [mailto:brian at freeswitch.org] Sent: 24 June 2009 19:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls left on FS after redirect offof FS I have tried to reproduce this issue but haven't been able too... What SVN Rev are you on? /b On Jun 24, 2009, at 10:29 AM, Richard Lamkin wrote: I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg call-command: execute execute-app-name: redirect execute-app-arg: sip:@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg call-command: execute execute-app-name: hangup execute-app-arg: Using command "api show channels" I find the following held on FS The only way I've found to remove these calls is "api hupall" ------------------------- uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196598 at 194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196599 at 194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728420 at 194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728400 at 194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137881 at 194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. ------------------- The SIP signalling is correct with an outgoing "302 moved temporarily" [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either "api hupall", or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to "api hupall". "api hupall" kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lamkin at mettoni.com ************************************************************************ * Please consider the environment before printing this e-mail ************************************************************************ * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/bc93458a/attachment-0002.html From brian at freeswitch.org Thu Jun 25 16:56:14 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 18:56:14 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins In-Reply-To: <24A1F18D-71EE-4751-967E-A450ED9185FF@freeswitch.org> References: <1384.1245405662@enterux.com> <24A1F18D-71EE-4751-967E-A450ED9185FF@freeswitch.org> Message-ID: <8B2560DD-4788-4823-A121-1C9A7C4C159A@freeswitch.org> Everyone that wanted to help this project to pay Arsen to write this please paypal brian at freeswitch.org when you can so I can gather it all up and send it to Arsen... Everyone that has sent money already thank you... ;) http://code.google.com/p/unimrcp/source/browse/trunk/plugins/mrcp-flite/src/mrcp_flite.c http://code.google.com/p/unimrcp/source/browse/trunk/plugins/mrcp-pocketsphinx/src/mrcp_pocketsphinx.c So the progress is moving forward.... Please pitch in. Thanks, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/6c8b9eb8/attachment-0002.html From danishmoosa at gmail.com Thu Jun 25 17:34:52 2009 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Fri, 26 Jun 2009 06:34:52 +0600 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? Message-ID: Hi Can somebody tell if FS freezes/crashes due to any reason. Does it logs both log of each call before dying? If we run it on large scale like 2-3k calls , a simple crash can cost a lot if it dies silently. One more aspect is , after freezing it will no more send/rec packets to any endpoint ,may result in inaccurate logging on endpoint. It should somehow send BYE ? -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/c3ee4b28/attachment-0002.html From brian at freeswitch.org Thu Jun 25 17:44:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 19:44:20 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: Message-ID: No the goal is to NOT crash in the first place. Are you experiencing a crash? If so http://wiki.freeswitch.org/wiki/Reporting_Bugs is how you would report it. Thanks, Brian On Jun 25, 2009, at 7:34 PM, Muhammad Danish Moosa wrote: > Hi > > Can somebody tell if FS freezes/crashes due to any reason. Does it > logs both log of each call before dying? > > If we run it on large scale like 2-3k calls , a simple crash can > cost a lot if it dies silently. One more aspect is , after freezing > it will no more send/rec packets to any endpoint ,may result in > inaccurate logging on endpoint. It should somehow send BYE ? > > -- > Muhammad Danish Moosa Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/1e03c0c3/attachment-0002.html From mgg at giagnocavo.net Thu Jun 25 18:16:13 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 25 Jun 2009 21:16:13 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> Well this isn't specific to FS crashing. The machine losing power would have the same effect, no? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Danish Moosa Sent: Thursday, June 25, 2009 6:35 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] CDR loss possibility if FS freezes? Hi Can somebody tell if FS freezes/crashes due to any reason. Does it logs both log of each call before dying? If we run it on large scale like 2-3k calls , a simple crash can cost a lot if it dies silently. One more aspect is , after freezing it will no more send/rec packets to any endpoint ,may result in inaccurate logging on endpoint. It should somehow send BYE ? -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/398e914a/attachment-0002.html From jingwei.yang at gmail.com Thu Jun 25 19:15:28 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 26 Jun 2009 10:15:28 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> Message-ID: <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> Hi Chris. thanks for the reply. Here're my answers. On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen wrote: > Two questions for you: > > 1) Do you have extension "888" in your public context? What public context are you saying? I only defined 888.xml in /usr/local/freeswitch/conf/directory/default. > 2)Can you put your internal Ip address of FS in rtp-ip instead of > $${bind_server_ip} just to make sure it get the right IP? I changed it to the internal Ip, but still no echo. > > 3) is not really > required at least for my working setup behind the NAT router. Thanks, I've commented it out. > > > > On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang wrote: > >> Hi Chris, thanks for your help. Here's my client.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: >> >>> Please provide your client.xml detail with confidential information >>> crossout, I have gtalk client and server working properly behind the NAT. >>> I should be able to help you. >>> >>> Chris >>> >>> >>> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: >>> >>>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >>>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >>>> audio this time no matter what value disable-rtp-auto-adjust is... >>>> >>>> >>>> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >>>> >>>>> uncomment ext-rtp-ip >>>>> >>>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>>>> >>>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and >>>>> put "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>>>> parameter has (true or false), local IP is able to hear the echo but >>>>> external ones still have no audio. >>>>> >>>>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>>>> >>>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember >>>>>> the exact var. >>>>>> >>>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>>>> >>>>>> Hi Guys, >>>>>> >>>>>> Here's my situation: >>>>>> >>>>>> The freeswitch server and my machine are behind the same LAN. If I >>>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>>>> >>>>>> However, external calls have no sound at all no matter whether this >>>>>> param is commented out or not. >>>>>> >>>>>> Please kindly let me know what other params to set to resolve this >>>>>> issue. >>>>>> >>>>>> Thanks, >>>>>> -Jingwei >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/0185e1fa/attachment-0002.html From chris.chen2004 at gmail.com Thu Jun 25 19:30:19 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Thu, 25 Jun 2009 22:30:19 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> Message-ID: <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> I guess you have the problem here, in client.xml you have but you only define extension 888 in default context, that's why nobody can reach you from public. under /usr/local/freeswitch/conf/dialplan define extension 888 in public.xml to the proper extension you expect, and check the console log from fs_cli when you do gtalk calling to your gmail client, you will find out the solution to your issue. chris On Thu, Jun 25, 2009 at 10:15 PM, Jingwei Yang wrote: > Hi Chris. thanks for the reply. Here're my answers. > > On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen wrote: > >> Two questions for you: >> >> 1) Do you have extension "888" in your public context? > > > What public context are you saying? I only defined 888.xml in > /usr/local/freeswitch/conf/directory/default. > > >> 2)Can you put your internal Ip address of FS in rtp-ip instead of >> $${bind_server_ip} just to make sure it get the right IP? > > > I changed it to the internal Ip, but still no echo. > > >> >> 3) is not really >> required at least for my working setup behind the NAT router. > > > Thanks, I've commented it out. > > >> >> >> >> On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang wrote: >> >>> Hi Chris, thanks for your help. Here's my client.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote: >>> >>>> Please provide your client.xml detail with confidential information >>>> crossout, I have gtalk client and server working properly behind the NAT. >>>> I should be able to help you. >>>> >>>> Chris >>>> >>>> >>>> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote: >>>> >>>>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip >>>>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no >>>>> audio this time no matter what value disable-rtp-auto-adjust is... >>>>> >>>>> >>>>> On Thu, Jun 25, 2009 at 11:24 AM, seven wrote: >>>>> >>>>>> uncomment ext-rtp-ip >>>>>> >>>>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: >>>>>> >>>>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and >>>>>> put "disable-rtp-auto-adjust" inside client.xml. No matter what value this >>>>>> parameter has (true or false), local IP is able to hear the echo but >>>>>> external ones still have no audio. >>>>>> >>>>>> On Wed, Jun 24, 2009 at 6:01 PM, seven wrote: >>>>>> >>>>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember >>>>>>> the exact var. >>>>>>> >>>>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote: >>>>>>> >>>>>>> Hi Guys, >>>>>>> >>>>>>> Here's my situation: >>>>>>> >>>>>>> The freeswitch server and my machine are behind the same LAN. If I >>>>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by >>>>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*). >>>>>>> >>>>>>> However, external calls have no sound at all no matter whether this >>>>>>> param is commented out or not. >>>>>>> >>>>>>> Please kindly let me know what other params to set to resolve this >>>>>>> issue. >>>>>>> >>>>>>> Thanks, >>>>>>> -Jingwei >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/d747d0f1/attachment-0002.html From dome at tel.co.th Thu Jun 25 19:38:45 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 09:38:45 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> <8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com> <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> Message-ID: <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> 2009/6/26 Michael Jerris : > I said to just add the set import=nibble_rate, your re-setting it for no > reason (and getting rid of the change that should have helped) by your > import=nibble_account line I test it agin. import work. nibble can see nibble_rate , nibble_account in channel but i can't change nibble heratbeat so nibble use default heartbeat. Dome C. > Mike > On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: > > Just test. > i use javascript > > ?????? session.execute("set", "import=nibble_rate"); > ?????? session.execute("set", "import=nibble_account"); > ?????? session.execute("bridge", > "{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=0838833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); > > when call connected nibble do nothing? i found heartbeat > > mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! > when call disconnect nibble update amont. > mod_nibblebill.c:478 Billing 16 secs > > I think nibble still not found variable channel. > > Let's me share more information > > I want to use nibblebill for callingcard. (i have develop billing by > myself). i plan to use javascript connect to ODBC > when customer call my script query balance and say. > and then i loop for get destination (my customer want to dial many number). > when i got number my script query > gateway from DB.? i have 3 route and order by cost. > First plan i use > session.execute("bridge", > "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/xxxx at provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/xxxx at provder2"); > i modify nibblebill for match provider with my billing. > this case still fail. > > now i try > > ??? if > (session.ready()){ > ??????? s = new > Session("{absolute_codec_string='GSM,G729'}sofia/external/xxx at provider1" > > } > ??? if > (s.ready()){ > ??????? session.execute("set", > "nibble_rate=2.5"); > ??????? session.execute("set", > "nibble_account="+acaller); > ??????? session.execute("set", > "hangup_after_bridge=false"); > ??????? session.execute("set", > "provider_id="+dialprovider_id[1]); > > bridge(session,s); > ??? } > > and check hangup cause before try other provider. > > > > Please guide me it's right way or not ? > > > Dome C. > > > 2009/6/26 Darren Schreiber >> >> Did this work? Would love an update on this error/issue. >> ________________________________ >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Wednesday, June 24, 2009 8:15 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >> >> try adding >> >> before the bridge and report back results. >> Mike >> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >> >> Dear All, >> >> Look like nibblebill does't work with multiple gatreway. >> I try >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> nibblebill not found nibble_rate >> >> But >> ??????? >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203.xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> Work fine >> >> What's difference from set application and []? ? >> >> Best Regards. >> Dome C. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Jun 25 19:40:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 21:40:04 -0500 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> Message-ID: I have been testing dingaling all day... I added switch_nat routines to poke holes in the nat if needed if you're behind upnp or nat-pmp. /b On Jun 25, 2009, at 9:30 PM, Chris Chen wrote: > I guess you have the problem here, > in client.xml you have > > > but you only define extension 888 in default context, > that's why nobody can reach you from public. > > under /usr/local/freeswitch/conf/dialplan > > define extension 888 in public.xml to the proper extension you > expect, and check the console log from fs_cli when you do gtalk > calling to your gmail client, you will find out the solution to your > issue. > > chris From brian at freeswitch.org Thu Jun 25 20:09:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:09:11 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> Message-ID: <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. /b On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: > Well this isn?t specific to FS crashing. The machine losing power > would have the same effect, no? > > -Michael > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/dbba1a79/attachment-0002.html From shannon at sacredhearts.us Thu Jun 25 20:32:37 2009 From: shannon at sacredhearts.us (Shannon) Date: Thu, 25 Jun 2009 22:32:37 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> Message-ID: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> In case of bad battery in APC, are cdr's logged prior to system failure? On Thursday, June 25, 2009, Brian West wrote: > true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. > /b > On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: > Well this isn?t specific to FS crashing. The machine losing power would have the same effect, no??-Michael > Brian Westbrian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com? > > > > > -- Shannon From jingwei.yang at gmail.com Thu Jun 25 20:34:36 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 26 Jun 2009 11:34:36 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> Message-ID: <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> Hi Chris, here's the one that confuses me. As far as I understand, the extension 888 defined in public.xml is for picking up incoming calls. It should have no influence on outgoing calls, right? If not, what is to write to fit my case? (originate dingaling/gmail.com/userAAA at gmail.com&bridge(dingaling/ gmail.com/userBBB at gmail.com), both userAAA and userBBB can be internal or external). Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm not quite sure what to include. So I make it very simple. Here are three relative parameters in client.xml: Still, I got no echo for internal Ip calls. Please let me know where goes wrong. Thanks, -Jingwei On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen wrote: > I guess you have the problem here, > in client.xml you have > > > but you only define extension 888 in default context, > that's why nobody can reach you from public. > > under /usr/local/freeswitch/conf/dialplan > > define extension 888 in public.xml to the proper extension you expect, and > check the console log from fs_cli when you do gtalk calling to your gmail > client, you will find out the solution to your issue. > > chris > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/5fe6d7e1/attachment-0002.html From brian at freeswitch.org Thu Jun 25 20:42:21 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:42:21 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> You could use something like nibble bill and at most loose the last interval of the call before its billed. You're going thru a lot of what if's ... You can't account for everything and you shouldn't have all your eggs in the same basket. /b On Jun 25, 2009, at 10:32 PM, Shannon wrote: > In case of bad battery in APC, are cdr's logged prior to system > failure? From mike at jerris.com Thu Jun 25 20:42:31 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 25 Jun 2009 23:42:31 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: <3BD9E49A-D794-4AB0-986D-082A54A39A3D@jerris.com> Of course not. This is why many do billing in icrements like mod_nibblebill does. Radius (although not yet with our module) and diamater both work this way and solve this issue. This in combination with session timers adress this and the hangup issue during a catastophic switch or network failure. On Jun 25, 2009, at 11:32 PM, Shannon wrote: > In case of bad battery in APC, are cdr's logged prior to system > failure? > > On Thursday, June 25, 2009, Brian West wrote: >> true dat... but again our goal is to not crash in the first >> place :P... nice APC can take care of the no power thing. >> /b >> On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: >> Well this isn?t specific to FS crashing. The machine losing power >> would have the same effect, no? -Michael >> Brian Westbrian at freeswitch.org >> -- Meet us at ClueCon! http://www.cluecon.com > > >> >> >> >> >> > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From harmeet at litatel.com Thu Jun 25 20:48:45 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Thu, 25 Jun 2009 23:48:45 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: Just make sure the power is always there! ...I know that some parts of the world this is not easy to achieve. Harmeet On Thu, Jun 25, 2009 at 11:32 PM, Shannon wrote: > In case of bad battery in APC, are cdr's logged prior to system failure? > > On Thursday, June 25, 2009, Brian West wrote: > > true dat... but again our goal is to not crash in the first place :P... > nice APC can take care of the no power thing. > > /b > > On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: > > Well this isn?t specific to FS crashing. The machine losing power would > have the same effect, no? -Michael > > Brian Westbrian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/f08effe1/attachment-0002.html From harmeet at litatel.com Thu Jun 25 20:53:13 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Thu, 25 Jun 2009 23:53:13 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> Message-ID: Does nibblebill update balances in real-time for each and every call? Does it do every second (micro/nano second?). How does it affect the performance vs if its done at end of call? I know that is not desirable for calling card applications. Harmeet On Thu, Jun 25, 2009 at 11:42 PM, Brian West wrote: > You could use something like nibble bill and at most loose the last > interval of the call before its billed. You're going thru a lot of > what if's ... You can't account for everything and you shouldn't have > all your eggs in the same basket. > > > > /b > > On Jun 25, 2009, at 10:32 PM, Shannon wrote: > > > In case of bad battery in APC, are cdr's logged prior to system > > failure? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/ffa73fa6/attachment-0002.html From brian at freeswitch.org Thu Jun 25 20:56:07 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:56:07 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> Message-ID: <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> Well you would do it every 20-60 seconds maybe... It would be silly to do it every micro/nano second... it would cost you more in cpu and you don't gain much. /b On Jun 25, 2009, at 10:53 PM, Harmeet Singh wrote: > Does nibblebill update balances in real-time for each and every > call? Does it do every second (micro/nano second?). How does it > affect the performance vs if its done at end of call? I know that is > not desirable for calling card applications. > > Harmeet From brian at freeswitch.org Thu Jun 25 20:56:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 22:56:46 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> Message-ID: <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> Is solar an option? ;) /b On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote: > Just make sure the power is always there! ...I know that some parts > of the world this is not easy to achieve. > > Harmeet From harmeet at litatel.com Thu Jun 25 21:06:26 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Fri, 26 Jun 2009 00:06:26 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> Message-ID: Can the interval be easily configures based on the destination? Like small interval for destinations with cost per minute > $1.00 and large intervals for cheaper destinations? Harmeet On Thu, Jun 25, 2009 at 11:56 PM, Brian West wrote: > Well you would do it every 20-60 seconds maybe... It would be silly to > do it every micro/nano second... it would cost you more in cpu and you > don't gain much. > > /b > > On Jun 25, 2009, at 10:53 PM, Harmeet Singh wrote: > > > Does nibblebill update balances in real-time for each and every > > call? Does it do every second (micro/nano second?). How does it > > affect the performance vs if its done at end of call? I know that is > > not desirable for calling card applications. > > > > Harmeet > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/a2798cb9/attachment-0002.html From sprice at gmail.com Thu Jun 25 21:07:03 2009 From: sprice at gmail.com (SP) Date: Thu, 25 Jun 2009 23:07:03 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> Message-ID: <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> clouds On Thu, Jun 25, 2009 at 22:56, Brian West wrote: > Is solar an option? ;) > > /b > > On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote: > > > Just make sure the power is always there! ...I know that some parts > > of the world this is not easy to achieve. > > > > Harmeet > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/0cb8a59d/attachment-0002.html From brian at freeswitch.org Thu Jun 25 21:11:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 23:11:15 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <35C5ADA1-5408-46D4-B04A-5233903A8B09@freeswitch.org> <7B85F3A3-19C2-4E64-BEF1-C5C1E9341EED@freeswitch.org> Message-ID: <92D78B64-4AB7-442B-8596-DDA0D54AA6B9@freeswitch.org> yes you could. why not check it out and set it up ... its rather powerful. /b On Jun 25, 2009, at 11:06 PM, Harmeet Singh wrote: > Can the interval be easily configures based on the destination? Like > small interval for destinations with cost per minute > $1.00 and > large intervals for cheaper destinations? > > Harmeet From brian at freeswitch.org Thu Jun 25 21:11:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 25 Jun 2009 23:11:31 -0500 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> Message-ID: <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> I also forgot about Nights. /b On Jun 25, 2009, at 11:07 PM, SP wrote: > clouds > > On Thu, Jun 25, 2009 at 22:56, Brian West > wrote: > Is solar an option? ;) > > /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090625/3f557c89/attachment-0002.html From intralanman at freeswitch.org Thu Jun 25 21:51:34 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 26 Jun 2009 00:51:34 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> Message-ID: <4A4453D6.4060107@freeswitch.org> windmills From darklion11 at yahoo.com Thu Jun 25 22:48:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 22:48:01 -0700 (PDT) Subject: [Freeswitch-users] multiple gateways not working? Message-ID: <24215324.post@talk.nabble.com> --> --> Is this correct for multiple gateways? When I try this the first gateway works but the second gateway does not work? What is the solution for this can u help me? Thanks -- View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dome at tel.co.th Thu Jun 25 23:13:44 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:13:44 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215324.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> Message-ID: <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> May be need before first bridge Dome C. 2009/6/26 Edmar Cruz : > > > ? > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? ?--> > ? ? ? > ? ? > > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? ?--> > ? ? ? > ? ? > > ? > > > > Is this correct for multiple gateways? When I try this the first gateway > works but the second gateway does not work? > > > What is the solution for this can u help me? > > > Thanks > > -- > View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Thu Jun 25 23:31:40 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 26 Jun 2009 16:31:40 +1000 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> Message-ID: <20090626063140.GA13942@jdc.jasonjgw.net> Dome Charoenyost wrote: > May be need > > before first bridge and also, reading this wiki page may help http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate (see the discussion of multiple destinations) From darklion11 at yahoo.com Thu Jun 25 23:34:06 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 25 Jun 2009 23:34:06 -0700 (PDT) Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> Message-ID: <24215631.post@talk.nabble.com> I try what you said still not working... Dome Charoenyost wrote: > > May be need > > before first bridge > > > Dome C. > 2009/6/26 Edmar Cruz : >> >> >> ? >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> ? ? ?--> >> ? ? ? >> ? ? >> >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> ? ? ?--> >> ? ? ? >> ? ? >> >> ? >> >> >> >> Is this correct for multiple gateways? When I try this the first gateway >> works but the second gateway does not work? >> >> >> What is the solution for this can u help me? >> >> >> Thanks >> >> -- >> View this message in context: >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dome at tel.co.th Thu Jun 25 23:45:02 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:45:02 +0700 Subject: [Freeswitch-users] From Asterisk to Freeswitch Message-ID: <8ccbff060906252345w1aa7c4a1u66f1c2af752dea20@mail.gmail.com> Dear All, I'm asterisk developer(I have some code in Asterisk) . After 3 weeks with freeswich nothing to say. now i'm move all callingcard , wholesale platfrom to Freeswitch. I'm very happy with bridge , nibblebill. after finish this job i'll test FS PBX feature. i think it's easy to do Hosted IPPBX. But i want to know more about mod_fifo Can someone tell me about mod_fifo compare with asterisk app_queue. i'm talking about annouce , priority agent Best Regards. Dome C. From dome at tel.co.th Thu Jun 25 23:46:14 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:46:14 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215631.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> Message-ID: <8ccbff060906252346r301469edg9350bc7091b12710@mail.gmail.com> Please try 2009/6/26 Edmar Cruz : > > > ? > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? > ? ? ? > ? ? > > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > > ? ? ? > ? ? > > ? > > > I try what you said still not working... > > > Dome Charoenyost wrote: >> >> May be need >> >> before first bridge >> >> >> Dome C. >> 2009/6/26 Edmar Cruz : >>> >>> >>> ? >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? >>> >>> >>> >>> Is this correct for multiple gateways? When I try this the first gateway >>> works but the second gateway does not work? >>> >>> >>> What is the solution for this can u help me? >>> >>> >>> Thanks >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dome at tel.co.th Thu Jun 25 23:48:03 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 13:48:03 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215631.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> Message-ID: <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> Or Try pipe if you want to ring all. Try comma 2009/6/26 Edmar Cruz : > > > ? > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > ? ? > ? ? ? > ? ? > > ? ? > ? ? ? > ? ? ? data="effective_caller_id_name=${effective_caller_id_name}"/> > ? ? ? data="effective_caller_id_number=${effective_caller_id_number}"/> > > ? ? ? > ? ? > > ? > > > I try what you said still not working... > > > Dome Charoenyost wrote: >> >> May be need >> >> before first bridge >> >> >> Dome C. >> 2009/6/26 Edmar Cruz : >>> >>> >>> ? >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? ? >>> ? ? ? >>> ? ? ?>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>> ? ? ?>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>> ? ? ?--> >>> ? ? ? >>> ? ? >>> >>> ? >>> >>> >>> >>> Is this correct for multiple gateways? When I try this the first gateway >>> works but the second gateway does not work? >>> >>> >>> What is the solution for this can u help me? >>> >>> >>> Thanks >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gcd at i.ph Thu Jun 25 23:49:26 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 26 Jun 2009 14:49:26 +0800 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215324.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> Message-ID: <7d0bfd8c0906252349ge0bf82dsa7f6c2250151f401@mail.gmail.com> you can combine the 2 gateways into one bridge app. pls see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall /nandy On Fri, Jun 26, 2009 at 1:48 PM, Edmar Cruz wrote: > > > > > > data="effective_caller_id_name=${effective_caller_id_name}"/> > data="effective_caller_id_number=${effective_caller_id_number}"/> > --> > > > > > > data="effective_caller_id_name=${effective_caller_id_name}"/> > data="effective_caller_id_number=${effective_caller_id_number}"/> > --> > > > > > > > > Is this correct for multiple gateways? When I try this the first gateway > works but the second gateway does not work? > > > What is the solution for this can u help me? > > > Thanks > > -- > View this message in context: > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/7b79db6e/attachment-0002.html From darklion11 at yahoo.com Fri Jun 26 00:05:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 26 Jun 2009 00:05:04 -0700 (PDT) Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> Message-ID: <24215893.post@talk.nabble.com> Yup your suggestions works... But I want my to have a prefix for the second bridge Dome Charoenyost wrote: > > Or Try pipe > > if you want to ring all. Try comma > > > > 2009/6/26 Edmar Cruz : >> >> >> ? >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> ? ? >> ? ? ? >> ? ? >> >> ? ? >> ? ? ? >> ? ? ?> data="effective_caller_id_name=${effective_caller_id_name}"/> >> ? ? ?> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >> ? ? ? >> ? ? >> >> ? >> >> >> I try what you said still not working... >> >> >> Dome Charoenyost wrote: >>> >>> May be need >>> >>> before first bridge >>> >>> >>> Dome C. >>> 2009/6/26 Edmar Cruz : >>>> >>>> >>>> ? >>>> ? ? >>>> ? ? ? >>>> ? ? ?>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>>> ? ? ?>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>>> ? ? ?--> >>>> ? ? ?>>> data="sofia/default/$1 at 116.80.80.101"/> >>>> ? ? >>>> >>>> ? ? >>>> ? ? ? >>>> ? ? ?>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >>>> ? ? ?>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >>>> ? ? ?--> >>>> ? ? ?>>> data="sofia/default/$1 at 116.80.80.102"/> >>>> ? ? >>>> >>>> ? >>>> >>>> >>>> >>>> Is this correct for multiple gateways? When I try this the first >>>> gateway >>>> works but the second gateway does not work? >>>> >>>> >>>> What is the solution for this can u help me? >>>> >>>> >>>> Thanks >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215893.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Fri Jun 26 02:03:31 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 26 Jun 2009 02:03:31 -0700 (PDT) Subject: [Freeswitch-users] Service_not_implemented for mobile phones? Message-ID: <24217116.post@talk.nabble.com> Hi, I receive an error message service not implemented sometimes when calling a mobile phone number but sometimes it works. What maybe rhe cause of this error? I already installed zfone, perfectly connect to two freeswitch and the one issue I got today is these can you help me guys? Thanks -- View this message in context: http://www.nabble.com/Service_not_implemented-for-mobile-phones--tp24217116p24217116.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tayeb.meftah at gmail.com Fri Jun 26 02:25:04 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 26 Jun 2009 09:25:04 +0000 Subject: [Freeswitch-users] Freeswitch Calling cart application Message-ID: <4A4493F0.2040400@gmail.com> hello all, please i need a open source / free calling cart application to use with my freeswitch cool anyone chare any application with me? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4190 (20090626) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From mail at willboyce.com Fri Jun 26 03:17:41 2009 From: mail at willboyce.com (Will Boyce) Date: Fri, 26 Jun 2009 05:17:41 -0500 (CDT) Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <20344659.541246011373348.JavaMail.SYSTEM@man-00108> Message-ID: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> [ Optimised G.729A 'Howlet' for FreeSWITCH & Asterisk ] Howler Technologies are proud to announce today the launch of the first fully indemnified G.729A solution for FreeSWITCH. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source telecoms platforms. The G.729A Howlet ships as a drop-in module for FreeSWITCH or Asterisk, and enables cost-effective transcoding of G.729 and G.729A calls to other codecs. It scales to more than 500 concurrent transcoded calls on a single quad core server, and is licensed on a per concurrent channel basis. You can choose from two licensing models - fixed server perpetual and annual floating, the latter allowing you to 'float' your licensed channels across multiple servers for ultimate flexibility. Our unique floating licenses means you enable G.729A across your infrastructure without the administrative overhead of managing per-server licenses, and at a fraction of the initial cost of fixed-server licenses. Howlets are available for purchase immediately, and start at just ?3.99/channnel with all patent holder royalties taken care of. Download your free trial today ! http://www.howlertech.com/products/howlets/ -- Enjoy, The Howler Team < support at howlertech.com > Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7099 fax: +44 207 099 7098 http://www.howlertech.com/ Registered in England & Wales, Company No. 06285634 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/5638c104/attachment-0002.html From jason at jasonjgw.net Fri Jun 26 03:33:49 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 26 Jun 2009 20:33:49 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090525011736.GA25198@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> Message-ID: <20090626103349.GA25435@jdc.jasonjgw.net> I can report that this problem (the failure of mod_portaudio.so to be linked properly) still persists as of revision 13970. The operating system is Debian Testing, and the difficulty began after upgrading from Libtool 1 to Libtool 2.2.6a. If anyone else can reproduce this or suggest a means of tracking down the cause, this would be much appreciated. The result of running ldd -r on mod_portaudio.so is as follows. It shows that most of the undefined symbols are from the Alsa library. I have also searched my build logs, (even with VERBOSE=1) but without locating any output that seems suspect. linux-vdso.so.1 => (0x00007fff279ff000) libm.so.6 => /lib/libm.so.6 (0x00007f261f415000) libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0x00007f261efd5000) libpthread.so.0 => /lib/libpthread.so.0 (0x00007f261edb9000) libc.so.6 => /lib/libc.so.6 (0x00007f261ea66000) libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f261e814000) libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f261e473000) libncurses.so.5 => /lib/libncurses.so.5 (0x00007f261e234000) libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f261df25000) libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f261dd09000) libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f261daa9000) /lib64/ld-linux-x86-64.so.2 (0x00007f261f8ee000) libdl.so.2 => /lib/libdl.so.2 (0x00007f261d8a5000) libz.so.1 => /usr/lib/libz.so.1 (0x00007f261d68d000) libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f261d484000) undefined symbol: crypt_r (/opt/freeswitch/lib/libfreeswitch.so.1) undefined symbol: clock_gettime (/opt/freeswitch/lib/libfreeswitch.so.1) undefined symbol: snd_config (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_period_size_min (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_channels (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_period_size_near (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_rate_near (./mod_portaudio.so) undefined symbol: clock_gettime (./mod_portaudio.so) undefined symbol: snd_pcm_poll_descriptors_revents (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_access (./mod_portaudio.so) undefined symbol: snd_pcm_format_size (./mod_portaudio.so) undefined symbol: snd_strerror (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_periods_min (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_rate (./mod_portaudio.so) undefined symbol: snd_pcm_info_set_device (./mod_portaudio.so) undefined symbol: snd_pcm_status (./mod_portaudio.so) undefined symbol: snd_pcm_close (./mod_portaudio.so) undefined symbol: snd_lib_error_set_handler (./mod_portaudio.so) undefined symbol: snd_config_get_id (./mod_portaudio.so) undefined symbol: snd_pcm_avail_update (./mod_portaudio.so) undefined symbol: snd_pcm_info_set_subdevice (./mod_portaudio.so) undefined symbol: snd_pcm_area_copy (./mod_portaudio.so) undefined symbol: snd_pcm_state (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_channels_min (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_period_size (./mod_portaudio.so) undefined symbol: snd_pcm_info_get_card (./mod_portaudio.so) undefined symbol: snd_pcm_poll_descriptors_count (./mod_portaudio.so) undefined symbol: snd_ctl_open (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_buffer_size_min (./mod_portaudio.so) undefined symbol: snd_config_update (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_channels_max (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_test_format (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_buffer_size_max (./mod_portaudio.so) undefined symbol: snd_ctl_card_info_get_name (./mod_portaudio.so) undefined symbol: snd_pcm_status_sizeof (./mod_portaudio.so) undefined symbol: snd_pcm_delay (./mod_portaudio.so) undefined symbol: snd_pcm_drain (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_tstamp (./mod_portaudio.so) undefined symbol: snd_pcm_start (./mod_portaudio.so) undefined symbol: snd_pcm_open (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_periods_integer (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_test_access (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_silence_threshold (./mod_portaudio.so) undefined symbol: snd_pcm_areas_silence (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_format (./mod_portaudio.so) undefined symbol: snd_ctl_close (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_xfer_align (./mod_portaudio.so) undefined symbol: snd_pcm_mmap_commit (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_sizeof (./mod_portaudio.so) undefined symbol: snd_config_update_free_global (./mod_portaudio.so) undefined symbol: snd_pcm_mmap_begin (./mod_portaudio.so) undefined symbol: snd_ctl_card_info (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_get_boundary (./mod_portaudio.so) undefined symbol: snd_ctl_pcm_next_device (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_periods_max (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params (./mod_portaudio.so) undefined symbol: snd_config_get_string (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_any (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_state (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_test_period_size (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_current (./mod_portaudio.so) undefined symbol: snd_pcm_link (./mod_portaudio.so) undefined symbol: snd_config_iterator_entry (./mod_portaudio.so) undefined symbol: snd_pcm_poll_descriptors (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_set_buffer_size_near (./mod_portaudio.so) undefined symbol: snd_pcm_info_sizeof (./mod_portaudio.so) undefined symbol: snd_ctl_pcm_info (./mod_portaudio.so) undefined symbol: snd_pcm_nonblock (./mod_portaudio.so) undefined symbol: snd_config_search (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_rate_numden (./mod_portaudio.so) undefined symbol: snd_ctl_card_info_sizeof (./mod_portaudio.so) undefined symbol: snd_pcm_drop (./mod_portaudio.so) undefined symbol: snd_config_iterator_end (./mod_portaudio.so) undefined symbol: snd_config_iterator_next (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_start_threshold (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_trigger_tstamp (./mod_portaudio.so) undefined symbol: snd_pcm_info_set_stream (./mod_portaudio.so) undefined symbol: snd_pcm_info (./mod_portaudio.so) undefined symbol: snd_config_iterator_first (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_stop_threshold (./mod_portaudio.so) undefined symbol: snd_card_next (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_silence_size (./mod_portaudio.so) undefined symbol: snd_pcm_status_get_delay (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_buffer_size (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_avail_min (./mod_portaudio.so) undefined symbol: snd_pcm_info_get_name (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params (./mod_portaudio.so) undefined symbol: snd_pcm_prepare (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_sizeof (./mod_portaudio.so) undefined symbol: snd_pcm_hw_params_get_period_size_max (./mod_portaudio.so) undefined symbol: snd_pcm_sw_params_set_tstamp_mode (./mod_portaudio.so) From jalsot at gmail.com Fri Jun 26 04:03:36 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 26 Jun 2009 13:03:36 +0200 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090626103349.GA25435@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> Message-ID: <4A44AB08.1050503@gmail.com> Did you make bootstrap.sh and configure before compilation? We have no issues on Ubuntu 9.04 (libtool 2.2.6a). Regards, Tamas Jason White ?rta: > I can report that this problem (the failure of mod_portaudio.so to be linked > properly) still persists as of revision 13970. > > The operating system is Debian Testing, and the difficulty began after > upgrading from Libtool 1 to Libtool 2.2.6a. > > If anyone else can reproduce this or suggest a means of tracking down the > cause, this would be much appreciated. > > The result of running ldd -r on mod_portaudio.so is as follows. It shows that > most of the undefined symbols are from the Alsa library. > > I have also searched my build logs, (even with VERBOSE=1) but without locating > any output that seems suspect. > > linux-vdso.so.1 => (0x00007fff279ff000) > libm.so.6 => /lib/libm.so.6 (0x00007f261f415000) > libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0x00007f261efd5000) > libpthread.so.0 => /lib/libpthread.so.0 (0x00007f261edb9000) > libc.so.6 => /lib/libc.so.6 (0x00007f261ea66000) > libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f261e814000) > libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f261e473000) > libncurses.so.5 => /lib/libncurses.so.5 (0x00007f261e234000) > libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f261df25000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f261dd09000) > libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f261daa9000) > /lib64/ld-linux-x86-64.so.2 (0x00007f261f8ee000) > libdl.so.2 => /lib/libdl.so.2 (0x00007f261d8a5000) > libz.so.1 => /usr/lib/libz.so.1 (0x00007f261d68d000) > libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f261d484000) > undefined symbol: crypt_r (/opt/freeswitch/lib/libfreeswitch.so.1) > undefined symbol: clock_gettime (/opt/freeswitch/lib/libfreeswitch.so.1) > undefined symbol: snd_config (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_period_size_min (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_channels (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_period_size_near (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_rate_near (./mod_portaudio.so) > undefined symbol: clock_gettime (./mod_portaudio.so) > undefined symbol: snd_pcm_poll_descriptors_revents (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_access (./mod_portaudio.so) > undefined symbol: snd_pcm_format_size (./mod_portaudio.so) > undefined symbol: snd_strerror (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_periods_min (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_rate (./mod_portaudio.so) > undefined symbol: snd_pcm_info_set_device (./mod_portaudio.so) > undefined symbol: snd_pcm_status (./mod_portaudio.so) > undefined symbol: snd_pcm_close (./mod_portaudio.so) > undefined symbol: snd_lib_error_set_handler (./mod_portaudio.so) > undefined symbol: snd_config_get_id (./mod_portaudio.so) > undefined symbol: snd_pcm_avail_update (./mod_portaudio.so) > undefined symbol: snd_pcm_info_set_subdevice (./mod_portaudio.so) > undefined symbol: snd_pcm_area_copy (./mod_portaudio.so) > undefined symbol: snd_pcm_state (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_channels_min (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_period_size (./mod_portaudio.so) > undefined symbol: snd_pcm_info_get_card (./mod_portaudio.so) > undefined symbol: snd_pcm_poll_descriptors_count (./mod_portaudio.so) > undefined symbol: snd_ctl_open (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_buffer_size_min (./mod_portaudio.so) > undefined symbol: snd_config_update (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_channels_max (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_test_format (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_buffer_size_max (./mod_portaudio.so) > undefined symbol: snd_ctl_card_info_get_name (./mod_portaudio.so) > undefined symbol: snd_pcm_status_sizeof (./mod_portaudio.so) > undefined symbol: snd_pcm_delay (./mod_portaudio.so) > undefined symbol: snd_pcm_drain (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_tstamp (./mod_portaudio.so) > undefined symbol: snd_pcm_start (./mod_portaudio.so) > undefined symbol: snd_pcm_open (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_periods_integer (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_test_access (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_silence_threshold (./mod_portaudio.so) > undefined symbol: snd_pcm_areas_silence (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_format (./mod_portaudio.so) > undefined symbol: snd_ctl_close (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_xfer_align (./mod_portaudio.so) > undefined symbol: snd_pcm_mmap_commit (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_sizeof (./mod_portaudio.so) > undefined symbol: snd_config_update_free_global (./mod_portaudio.so) > undefined symbol: snd_pcm_mmap_begin (./mod_portaudio.so) > undefined symbol: snd_ctl_card_info (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_get_boundary (./mod_portaudio.so) > undefined symbol: snd_ctl_pcm_next_device (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_periods_max (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params (./mod_portaudio.so) > undefined symbol: snd_config_get_string (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_any (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_state (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_test_period_size (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_current (./mod_portaudio.so) > undefined symbol: snd_pcm_link (./mod_portaudio.so) > undefined symbol: snd_config_iterator_entry (./mod_portaudio.so) > undefined symbol: snd_pcm_poll_descriptors (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_set_buffer_size_near (./mod_portaudio.so) > undefined symbol: snd_pcm_info_sizeof (./mod_portaudio.so) > undefined symbol: snd_ctl_pcm_info (./mod_portaudio.so) > undefined symbol: snd_pcm_nonblock (./mod_portaudio.so) > undefined symbol: snd_config_search (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_rate_numden (./mod_portaudio.so) > undefined symbol: snd_ctl_card_info_sizeof (./mod_portaudio.so) > undefined symbol: snd_pcm_drop (./mod_portaudio.so) > undefined symbol: snd_config_iterator_end (./mod_portaudio.so) > undefined symbol: snd_config_iterator_next (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_start_threshold (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_trigger_tstamp (./mod_portaudio.so) > undefined symbol: snd_pcm_info_set_stream (./mod_portaudio.so) > undefined symbol: snd_pcm_info (./mod_portaudio.so) > undefined symbol: snd_config_iterator_first (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_stop_threshold (./mod_portaudio.so) > undefined symbol: snd_card_next (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_silence_size (./mod_portaudio.so) > undefined symbol: snd_pcm_status_get_delay (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_buffer_size (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_avail_min (./mod_portaudio.so) > undefined symbol: snd_pcm_info_get_name (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params (./mod_portaudio.so) > undefined symbol: snd_pcm_prepare (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_sizeof (./mod_portaudio.so) > undefined symbol: snd_pcm_hw_params_get_period_size_max (./mod_portaudio.so) > undefined symbol: snd_pcm_sw_params_set_tstamp_mode (./mod_portaudio.so) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From odermann at googlemail.com Fri Jun 26 04:38:12 2009 From: odermann at googlemail.com (Dennis) Date: Fri, 26 Jun 2009 13:38:12 +0200 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud Message-ID: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> hi, we want to use uuid_displace with mux to playback a soundfile to a bridged uuid, so that this uuid can hear the other side talk AND hear the soundfile (whispering). is there an option we can set, for defining the loudness level of the soundfile? in our tests the soundfile was way to loud, so that it was nearly impossible to hear the other side talk, while the soundfile was playing. we tried "uuid_displace uuid start /path/to/soundfile/soundfile.wav 0 mux 0.3", so that the loudness of soundfile only would be 30% - but this does not work. thanks & kind regards dennis From harmeet at litatel.com Fri Jun 26 05:10:15 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Fri, 26 Jun 2009 08:10:15 -0400 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: <24215893.post@talk.nabble.com> References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> <24215893.post@talk.nabble.com> Message-ID: Just add the prefix like - BTW: Never give your real prefix. Anybody can use it to send traffic to your trunk with that prefix and eat away your balance. On Fri, Jun 26, 2009 at 3:05 AM, Edmar Cruz wrote: > > Yup your suggestions works... But I want my to have a prefix for the second > bridge > > > > > > Dome Charoenyost wrote: > > > > Or Try pipe > > > > if you want to ring all. Try comma > > > > > > > > 2009/6/26 Edmar Cruz : > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_name=${effective_caller_id_name}"/> > >> >> data="effective_caller_id_number=${effective_caller_id_number}"/> > >> > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_name=${effective_caller_id_name}"/> > >> >> data="effective_caller_id_number=${effective_caller_id_number}"/> > >> > >> > >> > >> > >> > >> > >> > >> I try what you said still not working... > >> > >> > >> Dome Charoenyost wrote: > >>> > >>> May be need > >>> > >>> before first bridge > >>> > >>> > >>> Dome C. > >>> 2009/6/26 Edmar Cruz : > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> > >>>> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> > >>>> --> > >>>> >>>> data="sofia/default/$1 at 116.80.80.101"/> > >>>> > >>>> > >>>> > >>>> > >>>> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> > >>>> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> > >>>> --> > >>>> >>>> data="sofia/default/$1 at 116.80.80.102"/> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> Is this correct for multiple gateways? When I try this the first > >>>> gateway > >>>> works but the second gateway does not work? > >>>> > >>>> > >>>> What is the solution for this can u help me? > >>>> > >>>> > >>>> Thanks > >>>> > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215893.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/e4a30246/attachment-0002.html From harmeet at litatel.com Fri Jun 26 05:23:04 2009 From: harmeet at litatel.com (Harmeet Singh) Date: Fri, 26 Jun 2009 08:23:04 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <4A4453D6.4060107@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> <4A4453D6.4060107@freeswitch.org> Message-ID: Pedals! For countries with cheap labour, this can give endless power! Each pedal connected to a dynamo. Workers come in shift after shift and keep pedalling! Its a solution for unemployement and electricity in those countries! Check this out - http://www.msnbc.msn.com/id/26430304 On Fri, Jun 26, 2009 at 12:51 AM, Raymond Chandler < intralanman at freeswitch.org> wrote: > windmills > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/744c2c73/attachment-0002.html From excelsio at gmx.net Fri Jun 26 05:33:37 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Fri, 26 Jun 2009 14:33:37 +0200 Subject: [Freeswitch-users] [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. Message-ID: <20090626123337.243300@gmx.net> [eap] EAP NAK [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. Hi, having a new voip pbx (OmniPCX Enterprise 9.0) from Alcatel-Lucent, I now try to setup 802.1x with the phones, an Alcatel-Lucent IP Touch 4028 EE. Freeradius is 2.x on a Debian 5.0. My first attempt was with MD5, which works without any problem. Next step is TLS, which works at 50%. Well, the client authentication of TLS works, but when I configure to do a server authentication within the IP phone?s setup, it fails. So, here is the output from radiusd -Xf when server authentication is not added within the IP phone?s setup: ============================================================= FreeRADIUS Version 2.1.6, for host i686-pc-linux-gnu, built on May 29 2009 at 15:54:08 Copyright (C) 1999-2009 The FreeRADIUS server project and contributors. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. You may redistribute copies of FreeRADIUS under the terms of the GNU General Public License v2. Starting - reading configuration files ... including configuration file /etc/raddb/radiusd.conf including configuration file /etc/raddb/proxy.conf including configuration file /etc/raddb/clients.conf including files in directory /etc/raddb/modules/ including configuration file /etc/raddb/modules/radutmp including configuration file /etc/raddb/modules/pap including configuration file /etc/raddb/modules/attr_rewrite including configuration file /etc/raddb/modules/perl including configuration file /etc/raddb/modules/expr including configuration file /etc/raddb/modules/smbpasswd including configuration file /etc/raddb/modules/otp including configuration file /etc/raddb/modules/acct_unique including configuration file /etc/raddb/modules/chap including configuration file /etc/raddb/modules/krb5 including configuration file /etc/raddb/modules/files including configuration file /etc/raddb/modules/policy including configuration file /etc/raddb/modules/wimax including configuration file /etc/raddb/modules/ldap including configuration file /etc/raddb/modules/realm including configuration file /etc/raddb/modules/mschap including configuration file /etc/raddb/modules/linelog including configuration file /etc/raddb/modules/sql_log including configuration file /etc/raddb/modules/detail including configuration file /etc/raddb/modules/digest including configuration file /etc/raddb/modules/mac2vlan including configuration file /etc/raddb/modules/expiration including configuration file /etc/raddb/modules/echo including configuration file /etc/raddb/modules/detail.example.com including configuration file /etc/raddb/modules/counter including configuration file /etc/raddb/modules/attr_filter including configuration file /etc/raddb/modules/sradutmp including configuration file /etc/raddb/modules/pam including configuration file /etc/raddb/modules/inner-eap including configuration file /etc/raddb/modules/smsotp including configuration file /etc/raddb/modules/passwd including configuration file /etc/raddb/modules/ippool including configuration file /etc/raddb/modules/logintime including configuration file /etc/raddb/modules/always including configuration file /etc/raddb/modules/preprocess including configuration file /etc/raddb/modules/sqlcounter_expire_on_login including configuration file /etc/raddb/modules/exec including configuration file /etc/raddb/modules/unix including configuration file /etc/raddb/modules/etc_group including configuration file /etc/raddb/modules/mac2ip including configuration file /etc/raddb/modules/checkval including configuration file /etc/raddb/modules/detail.log including configuration file /etc/raddb/eap.conf including configuration file /etc/raddb/policy.conf including files in directory /etc/raddb/sites-enabled/ including configuration file /etc/raddb/sites-enabled/inner-tunnel including configuration file /etc/raddb/sites-enabled/default including configuration file /etc/raddb/sites-enabled/control-socket including dictionary file /etc/raddb/dictionary main { prefix = "/usr" localstatedir = "/var" logdir = "/var/log/radius" libdir = "/usr/lib/freeradius" radacctdir = "/var/log/radius/radacct" hostname_lookups = no max_request_time = 30 cleanup_delay = 5 max_requests = 1024 allow_core_dumps = no pidfile = "/var/run/radiusd/radiusd.pid" checkrad = "/usr/sbin/checkrad" debug_level = 0 proxy_requests = yes log { stripped_names = no auth = no auth_badpass = no auth_goodpass = no } security { max_attributes = 200 reject_delay = 1 status_server = yes } } radiusd: #### Loading Realms and Home Servers #### proxy server { retry_delay = 5 retry_count = 3 default_fallback = no dead_time = 120 wake_all_if_all_dead = no } home_server localhost { ipaddr = 127.0.0.1 port = 1812 type = "auth" secret = "testing123" response_window = 20 max_outstanding = 65536 require_message_authenticator = no zombie_period = 40 status_check = "status-server" ping_interval = 30 check_interval = 30 num_answers_to_alive = 3 num_pings_to_alive = 3 revive_interval = 120 status_check_timeout = 4 irt = 2 mrt = 16 mrc = 5 mrd = 30 } home_server_pool my_auth_failover { type = fail-over home_server = localhost } realm example.com { auth_pool = my_auth_failover } realm LOCAL { } radiusd: #### Loading Clients #### client localhost { ipaddr = 127.0.0.1 require_message_authenticator = no secret = "testing123" nastype = "other" } client 192.168.10.130 { require_message_authenticator = no secret = "password" shortname = "Switch1" } client 192.168.10.131 { require_message_authenticator = no secret = "password" shortname = "Switch2" } client 192.168.104.244 { require_message_authenticator = no secret = "password" shortname = "Switch3" } radiusd: #### Instantiating modules #### instantiate { Module: Linked to module rlm_exec Module: Instantiating exec exec { wait = no input_pairs = "request" shell_escape = yes } Module: Linked to module rlm_expr Module: Instantiating expr Module: Linked to module rlm_expiration Module: Instantiating expiration expiration { reply-message = "Password Has Expired " } Module: Linked to module rlm_logintime Module: Instantiating logintime logintime { reply-message = "You are calling outside your allowed timespan " minimum-timeout = 60 } } radiusd: #### Loading Virtual Servers #### server inner-tunnel { modules { Module: Checking authenticate {...} for more modules to load Module: Linked to module rlm_pap Module: Instantiating pap pap { encryption_scheme = "auto" auto_header = no } Module: Linked to module rlm_chap Module: Instantiating chap Module: Linked to module rlm_mschap Module: Instantiating mschap mschap { use_mppe = yes require_encryption = no require_strong = no with_ntdomain_hack = no } Module: Linked to module rlm_unix Module: Instantiating unix unix { radwtmp = "/var/log/radius/radwtmp" } Module: Linked to module rlm_eap Module: Instantiating eap eap { default_eap_type = "tls" timer_expire = 60 ignore_unknown_eap_types = no cisco_accounting_username_bug = no max_sessions = 2048 } Module: Linked to sub-module rlm_eap_md5 Module: Instantiating eap-md5 Module: Linked to sub-module rlm_eap_leap Module: Instantiating eap-leap Module: Linked to sub-module rlm_eap_gtc Module: Instantiating eap-gtc gtc { challenge = "Password: " auth_type = "PAP" } Module: Linked to sub-module rlm_eap_tls Module: Instantiating eap-tls tls { rsa_key_exchange = yes dh_key_exchange = yes rsa_key_length = 2048 dh_key_length = 512 verify_depth = 0 pem_file_type = yes private_key_file = "/etc/raddb/certs/server.key" certificate_file = "/etc/raddb/certs/server.crt" CA_file = "/etc/raddb/certs/ca.crt" private_key_password = "password" dh_file = "/etc/raddb/certs/dh" random_file = "/etc/raddb/certs/random" fragment_size = 1024 include_length = yes check_crl = no cipher_list = "DEFAULT" make_cert_command = "/etc/raddb/certs/bootstrap" cache { enable = no lifetime = 24 max_entries = 255 } } Module: Linked to sub-module rlm_eap_ttls Module: Instantiating eap-ttls ttls { default_eap_type = "md5" copy_request_to_tunnel = yes use_tunneled_reply = yes virtual_server = "inner-tunnel" include_length = yes } Module: Linked to sub-module rlm_eap_peap Module: Instantiating eap-peap peap { default_eap_type = "mschapv2" copy_request_to_tunnel = no use_tunneled_reply = no proxy_tunneled_request_as_eap = yes virtual_server = "inner-tunnel" } Module: Linked to sub-module rlm_eap_mschapv2 Module: Instantiating eap-mschapv2 mschapv2 { with_ntdomain_hack = no } Module: Checking authorize {...} for more modules to load Module: Linked to module rlm_realm Module: Instantiating suffix realm suffix { format = "suffix" delimiter = "@" ignore_default = no ignore_null = no } Module: Linked to module rlm_files Module: Instantiating files files { usersfile = "/etc/raddb/users" acctusersfile = "/etc/raddb/acct_users" preproxy_usersfile = "/etc/raddb/preproxy_users" compat = "no" } Module: Checking session {...} for more modules to load Module: Linked to module rlm_radutmp Module: Instantiating radutmp radutmp { filename = "/var/log/radius/radutmp" username = "%{User-Name}" case_sensitive = yes check_with_nas = yes perm = 384 callerid = yes } Module: Checking post-proxy {...} for more modules to load Module: Checking post-auth {...} for more modules to load Module: Linked to module rlm_attr_filter Module: Instantiating attr_filter.access_reject attr_filter attr_filter.access_reject { attrsfile = "/etc/raddb/attrs.access_reject" key = "%{User-Name}" } } # modules } # server server { modules { Module: Checking authenticate {...} for more modules to load Module: Checking authorize {...} for more modules to load Module: Linked to module rlm_preprocess Module: Instantiating preprocess preprocess { huntgroups = "/etc/raddb/huntgroups" hints = "/etc/raddb/hints" with_ascend_hack = no ascend_channels_per_line = 23 with_ntdomain_hack = no with_specialix_jetstream_hack = no with_cisco_vsa_hack = no with_alvarion_vsa_hack = no } Module: Checking preacct {...} for more modules to load Module: Linked to module rlm_acct_unique Module: Instantiating acct_unique acct_unique { key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port" } Module: Checking accounting {...} for more modules to load Module: Linked to module rlm_detail Module: Instantiating detail detail { detailfile = "/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d" header = "%t" detailperm = 384 dirperm = 493 locking = no log_packet_header = no } Module: Instantiating attr_filter.accounting_response attr_filter attr_filter.accounting_response { attrsfile = "/etc/raddb/attrs.accounting_response" key = "%{User-Name}" } Module: Checking session {...} for more modules to load Module: Checking post-proxy {...} for more modules to load Module: Checking post-auth {...} for more modules to load } # modules } # server radiusd: #### Opening IP addresses and Ports #### listen { type = "auth" ipaddr = * port = 0 } listen { type = "acct" ipaddr = * port = 0 } listen { type = "control" listen { socket = "/var/run/radiusd/radiusd.sock" } } Listening on authentication address * port 1812 Listening on accounting address * port 1813 Listening on command file /var/run/radiusd/radiusd.sock Listening on proxy address * port 1814 Ready to process requests. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=127, length=336 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" EAP-Message = 0x0244001101303038303966383538633137 Message-Authenticator = 0x26f23798cfc04abb83fa87d210d1bc69 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 68 length 17 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop [pap] WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] EAP Identity [eap] processing type tls [tls] Requiring client certificate [tls] Initiate [tls] Start returned 1 ++[eap] returns handled Sending Access-Challenge of id 127 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x014500060d20 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aa8262fa7eff5a1519b312199 Finished request 0. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=128, length=437 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aa8262fa7eff5a1519b312199 EAP-Message = 0x024500640d800000005a160301005501000051030100000000029c3cefd84c43ff0fdd96f2139986e55316e45f34fe5e36a3caa07f00002a00660065006400630062006100600015001400120011000900080006000500040003001a0019001800170100 Message-Authenticator = 0xca28b57e66136938c479c4f7c87ca2b9 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 69 length 100 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS TLS Length 90 [tls] Length Included [tls] eaptls_verify returned 11 [tls] (other): before/accept initialization [tls] TLS_accept: before/accept initialization [tls] <<< TLS 1.0 Handshake [length 0055], ClientHello [tls] TLS_accept: SSLv3 read client hello A [tls] >>> TLS 1.0 Handshake [length 002a], ServerHello [tls] TLS_accept: SSLv3 write server hello A [tls] >>> TLS 1.0 Handshake [length 03e2], Certificate [tls] TLS_accept: SSLv3 write certificate A [tls] >>> TLS 1.0 Handshake [length 00c2], CertificateRequest [tls] TLS_accept: SSLv3 write certificate request A [tls] TLS_accept: SSLv3 flush data [tls] TLS_accept: Need to read more data: SSLv3 read client certificate A In SSL Handshake Phase In SSL Accept mode [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 128 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 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 EAP-Message = 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 EAP-Message = 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 EAP-Message = 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 EAP-Message = 0x710afb0e1495d38c42da7dd3 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aa9252fa7eff5a1519b312199 Finished request 1. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=129, length=343 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aa9252fa7eff5a1519b312199 EAP-Message = 0x024600060d00 Message-Authenticator = 0x5a31886be829f7fe67e6f24f7d17cd3f MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 70 length 6 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS [tls] Received TLS ACK [tls] ACK handshake fragment handler [tls] eaptls_verify returned 1 [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 129 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x014700f10d80000004dd8a8ab4bce56d175b1c0969bdd410271ff5999d0d1d3fb011bdfbe4504764c0b116030100c20d0000ba0301024000b400b23081af310b30090603550406130244453110300e0603550407130742616d62657267312c302a060355040a13234f74746f2d4672696564726963682d556e697665727369746165742042616d6265726731163014060355040b130d52656368656e7a656e7472756d311c301a060355040313134b6f6d6d756e696b6174696f6e736e65747a65312a302806092a864886f70d010901161b6e65747a2d7365727669636540756e692d62616d626572672e64650e000000 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aaa242fa7eff5a1519b312199 Finished request 2. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=130, length=1657 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aaa242fa7eff5a1519b312199 EAP-Message = 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 EAP-Message = 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 EAP-Message = 0x9912d806a75d89d3c89e227fe47bb1fd322cd39c97fab7290bbcef4e3af3d3710ca4a58b83bb8bc5edb4801d031c9da0136c646dc4c87c20cbb8ccf8578f89ba46679b1aba3b062528dd2f2591fb415012639bd24613cd1a91e6f2b5f0301e14092e85bde894b6633366492c9327f37e3b20a3a397e023749258f9c3d18bcaacfd03b0acf27f8c47dee5dd3dde4dd89a4a8ba166f7011d8ff95d2ef3b46af904100105be753df839c6f89e07dfea4ffb432781d9a4cc3c114f7b2b9d28026698c26caaabfeff0203010001300d06092a864886f70d0101050500038201010040bd27cdc3fe8ebbecbbabe6bbc6012d95325b282bd5d6141ef01de35a93 EAP-Message = 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 EAP-Message = 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 EAP-Message = 0xafb637c54c0f2af5ce0b900abaa00583a06ef6f041d6bdd639497450a2f527e4a0b53e41ee93ef1311eab231c6 Message-Authenticator = 0xa32aab76eea9d3033e8ea581033ddf23 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 71 length 253 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS TLS Length 1560 [tls] Received EAP-TLS First Fragment of the message [tls] eaptls_verify returned 9 [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 130 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x014800060d00 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aab2b2fa7eff5a1519b312199 Finished request 3. Going to the next request Waking up in 3.7 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=131, length=605 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aab2b2fa7eff5a1519b312199 EAP-Message = 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 EAP-Message = 0x14e12d8625cc5ce37e5e548243 Message-Authenticator = 0x5d76a81c99d9ed8d117658f6693d21f8 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 72 length 253 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS [tls] eaptls_verify returned 7 [tls] Done initial handshake [tls] <<< TLS 1.0 Handshake [length 03ca], Certificate [tls] chain-depth=1, [tls] error=0 [tls] --> User-Name = 00809f858c17 [tls] --> BUF-Name = [tls] --> subject = /C=DE/ [tls] --> issuer = /C=DE/ [tls] --> verify return:1 [tls] chain-depth=0, [tls] error=0 [tls] --> User-Name = 00809f858c17 [tls] --> BUF-Name = 3850 [tls] --> subject = /C=DE/L=[tls] --> issuer = /C=DE/L= [tls] --> verify return:1 [tls] TLS_accept: SSLv3 read client certificate A [tls] <<< TLS 1.0 Handshake [length 0106], ClientKeyExchange [tls] TLS_accept: SSLv3 read client key exchange A [tls] <<< TLS 1.0 Handshake [length 0106], CertificateVerify [tls] TLS_accept: SSLv3 read certificate verify A [tls] <<< TLS 1.0 ChangeCipherSpec [length 0001] [tls] <<< TLS 1.0 Handshake [length 0010], Finished [tls] TLS_accept: SSLv3 read finished A [tls] >>> TLS 1.0 ChangeCipherSpec [length 0001] [tls] TLS_accept: SSLv3 write change cipher spec A [tls] >>> TLS 1.0 Handshake [length 0010], Finished [tls] TLS_accept: SSLv3 write finished A [tls] TLS_accept: SSLv3 flush data [tls] (other): SSL negotiation finished successfully SSL Connection Established [tls] eaptls_process returned 13 ++[eap] returns handled Sending Access-Challenge of id 131 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x0149003d0d800000003314030100010116030100284751a33290290d3f84335c77caabe21228593b3c56db90e09d9bb1e672ef3a5285b5e7638d2931e5 Message-Authenticator = 0x00000000000000000000000000000000 State = 0xa863225aac2a2fa7eff5a1519b312199 Finished request 4. Going to the next request Waking up in 3.6 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=132, length=343 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0xa863225aac2a2fa7eff5a1519b312199 EAP-Message = 0x024900060d00 Message-Authenticator = 0xd9f31890976547a84f346a88f7c1d87f MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 73 length 6 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP/tls [eap] processing type tls [tls] Authenticate [tls] processing EAP-TLS [tls] Received TLS ACK [tls] ACK handshake is finished [tls] eaptls_verify returned 3 [tls] eaptls_process returned 3 [tls] Adding user data to cached session [eap] Freeing handler ++[eap] returns ok +- entering group post-auth {...} ++[exec] returns noop Sending Access-Accept of id 132 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP MS-MPPE-Recv-Key = 0x78b733f877a98c69f4197d25508c9528eb2cd0d68686ce27dffeb875d0550248 MS-MPPE-Send-Key = 0x5893ecdda559f6e70bafffb3088f4329608a138b74c62db3bf4eba8a187daf2d EAP-Message = 0x03490004 Message-Authenticator = 0x00000000000000000000000000000000 User-Name = "00809f858c17" Finished request 5. Going to the next request Waking up in 3.6 seconds. Cleaning up request 0 ID 127 with timestamp +2 Cleaning up request 1 ID 128 with timestamp +2 Cleaning up request 2 ID 129 with timestamp +2 Waking up in 1.2 seconds. Cleaning up request 3 ID 130 with timestamp +4 Cleaning up request 4 ID 131 with timestamp +4 Cleaning up request 5 ID 132 with timestamp +4 Ready to process requests. ============================================================================================================================= As soon as I enable "Server Authentication" wthin the IP phone, it fails: ============================================================================================================================= Going to the next request Ready to process requests. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=146, length=336 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" EAP-Message = 0x0217001101303038303966383538633137 Message-Authenticator = 0xb2b310ea9fb4000f3c2a436a6646b653 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 23 length 17 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop [pap] WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] EAP Identity [eap] processing type tls [tls] Requiring client certificate [tls] Initiate [tls] Start returned 1 ++[eap] returns handled Sending Access-Challenge of id 146 to 192.168.10.130 port 1812 Framed-Protocol = PPP Framed-Compression = Van-Jacobson-TCP-IP EAP-Message = 0x011800060d20 Message-Authenticator = 0x00000000000000000000000000000000 State = 0x750d139875151e34b18bc5eb9140ffe1 Finished request 9. Going to the next request Waking up in 4.9 seconds. rad_recv: Access-Request packet from host 192.168.10.130 port 1812, id=147, length=343 Framed-MTU = 1480 NAS-IP-Address = 192.168.10.130 NAS-Identifier = "Switch" User-Name = "00809f858c17" Service-Type = Framed-User Framed-Protocol = PPP NAS-Port = 199 NAS-Port-Type = Ethernet NAS-Port-Id = "I7" Called-Station-Id = "00-21-f7-eb-c3-00" Calling-Station-Id = "00-80-9f-85-8c-17" Connect-Info = "CONNECT Ethernet 1000Mbps Full duplex" Tunnel-Type:0 = VLAN Tunnel-Medium-Type:0 = IEEE-802 Tunnel-Private-Group-Id:0 = "104" State = 0x750d139875151e34b18bc5eb9140ffe1 EAP-Message = 0x02180006030d Message-Authenticator = 0xc861eecfeba7db7c00683c189e63b265 MS-RAS-Vendor = 11 HP-Attr-255 = 0x011a0000000b28 HP-Attr-255 = 0x011a0000000b2e HP-Attr-255 = 0x011a0000000b3d HP-Attr-255 = 0x0138 HP-Attr-255 = 0x013a HP-Attr-255 = 0x0140 HP-Attr-255 = 0x0141 HP-Attr-255 = 0x0151 +- entering group authorize {...} ++[preprocess] returns ok [suffix] No '@' in User-Name = "00809f858c17", looking up realm NULL [suffix] No such realm "NULL" ++[suffix] returns noop [eap] EAP packet type response id 24 length 6 [eap] No EAP Start, assuming it's an on-going EAP conversation ++[eap] returns updated ++[unix] returns notfound [files] users: Matched entry DEFAULT at line 234 ++[files] returns ok ++[expiration] returns noop ++[logintime] returns noop [pap] WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop Found Auth-Type = EAP +- entering group authenticate {...} [eap] Request found, released from the list [eap] EAP NAK [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. [eap] No common EAP types found. [eap] Failed in EAP select ++[eap] returns invalid Failed to authenticate the user. Using Post-Auth-Type Reject +- entering group REJECT {...} [attr_filter.access_reject] expand: %{User-Name} -> 00809f858c17 attr_filter: Matched entry DEFAULT at line 11 ++[attr_filter.access_reject] returns updated Delaying reject of request 10 for 1 seconds Going to the next request Waking up in 0.9 seconds. Sending delayed reject for request 10 Sending Access-Reject of id 147 to 192.168.10.130 port 1812 EAP-Message = 0x04180004 Message-Authenticator = 0x00000000000000000000000000000000 Waking up in 3.9 seconds. Cleaning up request 9 ID 146 with timestamp +131 Waking up in 1.0 seconds. Cleaning up request 10 ID 147 with timestamp +131 Ready to process requests. Well, what?s going wrong? Michael From excelsio at gmx.net Fri Jun 26 05:38:33 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Fri, 26 Jun 2009 14:38:33 +0200 Subject: [Freeswitch-users] [eap] ERROR! Our request for tls was NAK'd with a request for tls. Skipping the requested type. In-Reply-To: <20090626123337.243300@gmx.net> References: <20090626123337.243300@gmx.net> Message-ID: <20090626123833.103370@gmx.net> Sorry, wrong group :-) Freeswitch, Freeradius, both free sorry :-) From woof at iwoof.org Fri Jun 26 06:58:53 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 26 Jun 2009 09:58:53 -0400 Subject: [Freeswitch-users] CDR loss possibility if FS freezes? In-Reply-To: <4A4453D6.4060107@freeswitch.org> References: <6E8D2069C08AA84A83D336E996AE4C67027A965391@mse17be1.mse17.exchange.ms> <23509722-B44B-421E-8526-946C5940DD94@freeswitch.org> <7e2ac3270906252032k64699a7ay2617302d832c1c23@mail.gmail.com> <0129F691-CD3F-446E-98E2-79E0B62A0FAF@freeswitch.org> <7e2ac3270906252107j76f23dedveb064bfac4e0c40c@mail.gmail.com> <1A51CB2C-F070-4DDA-91F9-4C8CCA684F8F@freeswitch.org> <4A4453D6.4060107@freeswitch.org> Message-ID: On Fri, 26 Jun 2009 00:51:34 -0400, Raymond Chandler wrote: > windmills Just set them up to be right next to the output of the system's fan! I gotta get to the patent office, quick! --Woof! From intralanman at freeswitch.org Fri Jun 26 07:12:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 26 Jun 2009 10:12:15 -0400 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> Message-ID: <4A44D73F.2010408@freeswitch.org> There doesn't seem to be any direct link between FreeSWITCH and Howler. How does this benefit the project? Can we assume that Howler is "giving something back"? -Ray From dave at 3c.co.uk Fri Jun 26 07:38:43 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 26 Jun 2009 17:38:43 +0300 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <4A44D73F.2010408@freeswitch.org> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> Message-ID: <1246027123.4232.16.camel@dk-d820> > There doesn't seem to be any direct link between FreeSWITCH and Howler. > How does this benefit the project? Can we assume that Howler is "giving > something back"? Well, they're providing additional functionality which a lot of folk want, which requires substantial investment in money (visit sipro.com and have a look at the up-front license fees if you wish) and time (to get the thing working across the various platforms with the required licensing) and they've presumably taken a commercial risk so to do. Hats off to them: they've taken a risk, and the project benefits immensely from it. Expecting them to "give something back" in addition would be entirely unreasonable. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From d at d-man.org Fri Jun 26 08:01:38 2009 From: d at d-man.org (Darren Schreiber) Date: Fri, 26 Jun 2009 08:01:38 -0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com><22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com><8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com><39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> Message-ID: I can add a field to allow you to adjust the heartbeat on either channel if that's necessary. Right now you are right, it's using the global setting always. Is this important? -----Original Message----- From: Dome Charoenyost [mailto:dome at tel.co.th] Sent: Thursday, June 25, 2009 7:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway 2009/6/26 Michael Jerris : > I said to just add the set import=nibble_rate, your re-setting it for > no reason (and getting rid of the change that should have helped) by > your import=nibble_account line I test it agin. import work. nibble can see nibble_rate , nibble_account in channel but i can't change nibble heratbeat so nibble use default heartbeat. Dome C. > Mike > On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: > > Just test. > i use javascript > > ?????? session.execute("set", "import=nibble_rate"); > ?????? session.execute("set", "import=nibble_account"); > ?????? session.execute("bridge", > "{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=083 > 8833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); > > when call connected nibble do nothing? i found heartbeat > > mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! > when call disconnect nibble update amont. > mod_nibblebill.c:478 Billing 16 secs > > I think nibble still not found variable channel. > > Let's me share more information > > I want to use nibblebill for callingcard. (i have develop billing by > myself). i plan to use javascript connect to ODBC when customer call > my script query balance and say. > and then i loop for get destination (my customer want to dial many number). > when i got number my script query > gateway from DB.? i have 3 route and order by cost. > First plan i use > session.execute("bridge", > "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/xxxx > @provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/exte > rnal/xxxx at provder2"); i modify nibblebill for match provider with my > billing. > this case still fail. > > now i try > > ??? if > (session.ready()){ > ??????? s = new > Session("{absolute_codec_string='GSM,G729'}sofia/external/xxx at provider1" > > } > ??? if > (s.ready()){ > ??????? session.execute("set", > "nibble_rate=2.5"); > ??????? session.execute("set", > "nibble_account="+acaller); > ??????? session.execute("set", > "hangup_after_bridge=false"); > ??????? session.execute("set", > "provider_id="+dialprovider_id[1]); > > bridge(session,s); > ??? } > > and check hangup cause before try other provider. > > > > Please guide me it's right way or not ? > > > Dome C. > > > 2009/6/26 Darren Schreiber >> >> Did this work? Would love an update on this error/issue. >> ________________________________ >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Wednesday, June 24, 2009 8:15 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >> >> try adding >> before the >> bridge and report back results. >> Mike >> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >> >> Dear All, >> >> Look like nibblebill does't work with multiple gatreway. >> I try >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/extern >> al/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734 >> 000 at 202.xxx.xxx.xxx> >> >> nibblebill not found nibble_rate >> >> But >> ??????? >> ??????? > data="nibble_account=0838833133"/> >> ??????? > data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203 >> .xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >> >> Work fine >> >> What's difference from set application and []? ? >> >> Best Regards. >> Dome C. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Jun 26 08:08:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 26 Jun 2009 10:08:45 -0500 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <191c3a030906260802h790535cal4009d32225abee2f@mail.gmail.com> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> <1246027123.4232.16.camel@dk-d820> <191c3a030906260802h790535cal4009d32225abee2f@mail.gmail.com> Message-ID: <191c3a030906260808n684800afj7bbd8258354195c7@mail.gmail.com> Congrats to them for taking the risk. Just as an aside, just to let everyone know, I am taking the same risk. We have invested in the same deal from sipro and will shortly be offering an official FreeSWITCH g729 module in the very near future. This, of course, does benifit the project directly, because it gives a revenue source to help fund the project where naturally a 3rd party module mostly benifits users of FreeSWITCH which is, of course, not a bad thing. We might have had it sooner but we've been very busy making FreeSWITCH itself and it's a matter of first things first. So, again congrats guys, I hope your hardware codec boards sell and you leave a little room for us on the software side. On Jun 26, 2009 9:48 AM, "David Knell" wrote: > There doesn't seem to be any direct link between FreeSWITCH and Howler. > How does this benefit ... Well, they're providing additional functionality which a lot of folk want, which requires substantial investment in money (visit sipro.com and have a look at the up-front license fees if you wish) and time (to get the thing working across the various platforms with the required licensing) and they've presumably taken a commercial risk so to do. Hats off to them: they've taken a risk, and the project benefits immensely from it. Expecting them to "give something back" in addition would be entirely unreasonable. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lis... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/ddabec74/attachment-0002.html From brian at freeswitch.org Fri Jun 26 08:13:32 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 10:13:32 -0500 Subject: [Freeswitch-users] Bug reports Message-ID: FreeSWITCHers, We have written an extensive guide on posting bugs to jira. Over the past few weeks everyone has been a little lax in posting the correct info the first time. The questions we ask are NOT optional to fill out.. You must fill out every question on the list and "doesn't apply" or "n/a" are not acceptable answers. So help us help you fix your issues that you report. Remember provide all the details even if you think its not relevant to the issue at hand because even the smallest detail can help me reproduce your issue or result in me pulling my hair out and going crazy. :P http://wiki.freeswitch.org/wiki/Reporting_Bugs Also I would like to solicit volunteers to help manage, track and collect info for any Jira's please contact me... I have asked a few times but very few expressed interest in helping. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/9d17c269/attachment-0002.html From vhatz at kinetix.gr Fri Jun 26 08:20:22 2009 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Fri, 26 Jun 2009 18:20:22 +0300 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <1246027123.4232.16.camel@dk-d820> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> <1246027123.4232.16.camel@dk-d820> Message-ID: <4A44E736.9080501@kinetix.gr> I have to agree with David here. A G729 codec benefits the project immensely as it allows Freeswitch users to use the most widespread commercial codec. Best regards, Vlasis Hatzistavrou Kinetix Tele.com Hellas Ltd. Monastiriou 9 & Enotikon 54627 Thessaloniki Greece Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: vhatz at kinetix.gr http://www.kinetixtele.com David Knell wrote: >> There doesn't seem to be any direct link between FreeSWITCH and Howler. >> How does this benefit the project? Can we assume that Howler is "giving >> something back"? > > Well, they're providing additional functionality which a lot of folk > want, which requires substantial investment in money (visit sipro.com > and have a look at the up-front license fees if you wish) and time (to > get the thing working across the various platforms with the required > licensing) and they've presumably taken a commercial risk so to do. > > Hats off to them: they've taken a risk, and the project benefits > immensely from it. Expecting them to "give something back" in addition > would be entirely unreasonable. > > --Dave > From chris.chen2004 at gmail.com Fri Jun 26 08:26:16 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 26 Jun 2009 11:26:16 -0400 Subject: [Freeswitch-users] Bug reports In-Reply-To: References: Message-ID: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> Brian, I would like to be one of the volunteers helping to report issues. Chris On Fri, Jun 26, 2009 at 11:13 AM, Brian West wrote: > FreeSWITCHers, > > We have written an extensive guide on posting bugs to jira. Over the past > few weeks everyone has been a little lax in posting the correct info the > first time. The questions we ask are NOT optional to fill out.. You must > fill out every question on the list and "doesn't apply" or "n/a" are not > acceptable answers. So help us help you fix your issues that you report. > Remember provide all the details even if you think its not relevant to the > issue at hand because even the smallest detail can help me reproduce your > issue or result in me pulling my hair out and going crazy. :P > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also I would like to solicit volunteers to help manage, track and collect > info for any Jira's please contact me... I have asked a few times but very > few expressed interest in helping. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/6847f685/attachment-0002.html From dome at tel.co.th Fri Jun 26 08:28:07 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 22:28:07 +0700 Subject: [Freeswitch-users] multiple gateways not working? In-Reply-To: References: <24215324.post@talk.nabble.com> <8ccbff060906252313v5951184chb3f65cdd3ef28155@mail.gmail.com> <24215631.post@talk.nabble.com> <8ccbff060906252348i6301f220u61c848da3ae07fcf@mail.gmail.com> <24215893.post@talk.nabble.com> Message-ID: <8ccbff060906260828s368ba86dp38bd8f25c99e6f3e@mail.gmail.com> 2009/6/26 Harmeet Singh : > Just add the prefix like - > > > > BTW: Never give your real prefix. Anybody can use it to send traffic to your > trunk with that prefix and eat away your balance. ACL can help :) > > On Fri, Jun 26, 2009 at 3:05 AM, Edmar Cruz wrote: >> >> Yup your suggestions works... But I want my to have a prefix for the >> second >> bridge >> >> >> >> >> >> Dome Charoenyost wrote: >> > >> > Or Try pipe >> > ? >> > if you want to ring all. Try comma >> > ? >> > >> > >> > 2009/6/26 Edmar Cruz : >> >> >> >> >> >> ? >> >> ? ? >> >> ? ? ? >> >> ? ? ?> >> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >> ? ? ?> >> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >> ? ? >> >> ? ? ?> >> data="sofia/default/$1 at 116.80.80.101"/> >> >> ? ? >> >> >> >> ? ? >> >> ? ? ? >> >> ? ? ?> >> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >> ? ? ?> >> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >> >> >> ? ? ?> >> data="sofia/default/$1 at 116.80.80.102"/> >> >> ? ? >> >> >> >> ? >> >> >> >> >> >> I try what you said still not working... >> >> >> >> >> >> Dome Charoenyost wrote: >> >>> >> >>> May be need >> >>> >> >>> before first bridge >> >>> >> >>> >> >>> Dome C. >> >>> 2009/6/26 Edmar Cruz : >> >>>> >> >>>> >> >>>> ? >> >>>> ? ? >> >>>> ? ? ? >> >>>> ? ? ?> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >>>> ? ? ?> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >>>> ? ? ?--> >> >>>> ? ? ?> >>>> data="sofia/default/$1 at 116.80.80.101"/> >> >>>> ? ? >> >>>> >> >>>> ? ? >> >>>> ? ? ? >> >>>> ? ? ?> >>>> data="effective_caller_id_name=${effective_caller_id_name}"/> >> >>>> ? ? ?> >>>> data="effective_caller_id_number=${effective_caller_id_number}"/> >> >>>> ? ? ?--> >> >>>> ? ? ?> >>>> data="sofia/default/$1 at 116.80.80.102"/> >> >>>> ? ? >> >>>> >> >>>> ? >> >>>> >> >>>> >> >>>> >> >>>> Is this correct for multiple gateways? When I try this the first >> >>>> gateway >> >>>> works but the second gateway does not work? >> >>>> >> >>>> >> >>>> What is the solution for this can u help me? >> >>>> >> >>>> >> >>>> Thanks >> >>>> >> >>>> -- >> >>>> View this message in context: >> >>>> >> >>>> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215324.html >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> Freeswitch-users mailing list >> >>>> Freeswitch-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> _______________________________________________ >> >>> Freeswitch-users mailing list >> >>> Freeswitch-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >>> >> >> >> >> -- >> >> View this message in context: >> >> >> >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215631.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/multiple-gateways-not-working--tp24215324p24215893.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dome at tel.co.th Fri Jun 26 08:26:59 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 26 Jun 2009 22:26:59 +0700 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> <8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com> <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> Message-ID: <8ccbff060906260826g46c565c7sbe820d0b60cb7dd1@mail.gmail.com> 2009/6/26 Darren Schreiber : > I can add a field to allow you to adjust the heartbeat on either channel if > that's necessary. Right now you are right, it's using the global setting > always. > > Is this important? Yes important for me. and early media also :) 90% mobile in thailand use Music Ringback Tone service. if i disable ignore early media caller happy (heard music) but nibblebill start counting ;( Dome C. > > -----Original Message----- > From: Dome Charoenyost [mailto:dome at tel.co.th] > Sent: Thursday, June 25, 2009 7:39 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway > > 2009/6/26 Michael Jerris : >> I said to just add the set import=nibble_rate, your re-setting it for >> no reason (and getting rid of the change that should have helped) by >> your import=nibble_account line > I test it agin. > import work. ?nibble can see nibble_rate , nibble_account in channel but ?i > can't ?change nibble heratbeat ?so nibble use default heartbeat. > > > Dome C. > >> Mike >> On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: >> >> Just test. >> i use javascript >> >> ?????? session.execute("set", "import=nibble_rate"); >> ?????? session.execute("set", "import=nibble_account"); >> ?????? session.execute("bridge", >> "{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=083 >> 8833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); >> >> when call connected nibble do nothing? i found heartbeat >> >> mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! >> when call disconnect nibble update amont. >> mod_nibblebill.c:478 Billing 16 secs >> >> I think nibble still not found variable channel. >> >> Let's me share more information >> >> I want to use nibblebill for callingcard. (i have develop billing by >> myself). i plan to use javascript connect to ODBC when customer call >> my script query balance and say. >> and then i loop for get destination (my customer want to dial many > number). >> when i got number my script query >> gateway from DB.? i have 3 route and order by cost. >> First plan i use >> session.execute("bridge", >> "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/xxxx >> @provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/exte >> rnal/xxxx at provder2"); i modify nibblebill for match provider with my >> billing. >> this case still fail. >> >> now i try >> >> ??? if >> (session.ready()){ >> ??????? s = new >> Session("{absolute_codec_string='GSM,G729'}sofia/external/xxx at provider1" >> >> } >> ??? if >> (s.ready()){ >> ??????? session.execute("set", >> "nibble_rate=2.5"); >> ??????? session.execute("set", >> "nibble_account="+acaller); >> ??????? session.execute("set", >> "hangup_after_bridge=false"); >> ??????? session.execute("set", >> "provider_id="+dialprovider_id[1]); >> >> bridge(session,s); >> ??? } >> >> and check hangup cause before try other provider. >> >> >> >> Please guide me it's right way or not ? >> >> >> Dome C. >> >> >> 2009/6/26 Darren Schreiber >>> >>> Did this work? Would love an update on this error/issue. >>> ________________________________ >>> From: Michael Jerris [mailto:mike at jerris.com] >>> Sent: Wednesday, June 24, 2009 8:15 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >>> >>> try adding >>> before the >>> bridge and report back results. >>> Mike >>> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >>> >>> Dear All, >>> >>> Look like nibblebill does't work with multiple gatreway. >>> I try >>> ??????? >> data="nibble_account=0838833133"/> >>> ??????? >> data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/extern >>> al/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734 >>> 000 at 202.xxx.xxx.xxx> >>> >>> nibblebill not found nibble_rate >>> >>> But >>> ??????? >>> ??????? >> data="nibble_account=0838833133"/> >>> ??????? >> data="{absolute_codec_string='GSM,G729'}sofia/external/6626734000 at 203 >>> .xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >>> >>> Work fine >>> >>> What's difference from set application and []? ? >>> >>> Best Regards. >>> Dome C. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jay.fenton at howlertech.com Fri Jun 26 08:28:13 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Fri, 26 Jun 2009 17:28:13 +0200 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <1246027123.4232.16.camel@dk-d820> References: <32327494.561246011408145.JavaMail.SYSTEM@man-00108> <4A44D73F.2010408@freeswitch.org> <1246027123.4232.16.camel@dk-d820> Message-ID: <2825AB6A-55CB-4C19-A5E3-67D3A865D686@howlertech.com> Hey David, >> There doesn't seem to be any direct link between FreeSWITCH and >> Howler. >> How does this benefit the project? Can we assume that Howler is >> "giving >> something back"? > > Well, they're providing additional functionality which a lot of folk > want, which requires substantial investment in money (visit sipro.com > and have a look at the up-front license fees if you wish) and time (to > get the thing working across the various platforms with the required > licensing) and they've presumably taken a commercial risk so to do. > > Hats off to them: they've taken a risk, and the project benefits > immensely from it. Expecting them to "give something back" in > addition > would be entirely unreasonable. You're spot on on the somewhat extortionate up-front (and continuing royalty) fees, and there is substantial risk in us doing this which we hope will be mitigated by the rising interest in FreeSWITCH as a whole (a project which we've been huge evangelists of since the beginning). We hope that adding patent encumbered codecs like G.729 to FreeSWITCH will ultimately benefit the project and cause more people to be able to transition towards it from other platforms, building on the fantastic platform that Anthony and the rest of the gang have somehow managed to develop for free! Our initial feedback has been very promising, with a number of customers at least in the UK (including one major VoIP provider) that we've spoken to directly now seriously considering a migration to FreeSWITCH. We hope that the small per channel charge is not too onerous - we've tried to make it as cheap as possible by introducing the floating licenses (and including all support and maintenance upgrades free of charge), but there are very real costs here in royalties and development - how you guys do it mostly for free is beyond me! We have, of course, spoken to the FreeSWITCH team and offered to collaborate on this project (as they've also been working on this for a while) and we hope that we can join forces on it soon so that everyone benefits :) -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From mike at jerris.com Fri Jun 26 08:42:50 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 11:42:50 -0400 Subject: [Freeswitch-users] Nibblebill and multiple gateway In-Reply-To: <8ccbff060906260826g46c565c7sbe820d0b60cb7dd1@mail.gmail.com> References: <8ccbff060906232236rde26be4qcd44a5c3f62d75c@mail.gmail.com> <22CF668C-B3B0-400E-A66F-533A830FD951@jerris.com> <8ccbff060906251104y375ee87cid56a402f87a8724d@mail.gmail.com> <39E54E63-FAD6-4851-9B46-95E9F4EF953D@jerris.com> <8ccbff060906251938h23a652a9sc12c9bc319542962@mail.gmail.com> <8ccbff060906260826g46c565c7sbe820d0b60cb7dd1@mail.gmail.com> Message-ID: <63B6DBF6-C57B-43CB-893B-73EA6EC8BC57@jerris.com> if you want to import multiple vars you can set import to a comma separated list of vars. This should solve the problem in the short term, the question for Darren is, do you think we should make the mod read from the a or b channel for those vars. Mike On Jun 26, 2009, at 11:26 AM, Dome Charoenyost wrote: > 2009/6/26 Darren Schreiber : >> I can add a field to allow you to adjust the heartbeat on either >> channel if >> that's necessary. Right now you are right, it's using the global >> setting >> always. >> >> Is this important? > Yes important for me. and early media also :) > > 90% mobile in thailand use Music Ringback Tone service. if i disable > ignore early media caller happy (heard music) > but nibblebill start counting ;( > > > > > Dome C. > > >> >> -----Original Message----- >> From: Dome Charoenyost [mailto:dome at tel.co.th] >> Sent: Thursday, June 25, 2009 7:39 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >> >> 2009/6/26 Michael Jerris : >>> I said to just add the set import=nibble_rate, your re-setting it >>> for >>> no reason (and getting rid of the change that should have helped) by >>> your import=nibble_account line >> I test it agin. >> import work. nibble can see nibble_rate , nibble_account in >> channel but i >> can't change nibble heratbeat so nibble use default heartbeat. >> >> >> Dome C. >> >>> Mike >>> On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote: >>> >>> Just test. >>> i use javascript >>> >>> session.execute("set", "import=nibble_rate"); >>> session.execute("set", "import=nibble_account"); >>> session.execute("bridge", >>> "{absolute_codec_string='GSM,G729'} >>> [nibble_rate=0.5,nibble_account=083 >>> 8833133]sofia/external/xxxx at xxxx.xxx.xxx.xx"); >>> >>> when call connected nibble do nothing i found heartbeat >>> >>> mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT! >>> when call disconnect nibble update amont. >>> mod_nibblebill.c:478 Billing 16 secs >>> >>> I think nibble still not found variable channel. >>> >>> Let's me share more information >>> >>> I want to use nibblebill for callingcard. (i have develop billing by >>> myself). i plan to use javascript connect to ODBC when customer call >>> my script query balance and say. >>> and then i loop for get destination (my customer want to dial many >> number). >>> when i got number my script query >>> gateway from DB. i have 3 route and order by cost. >>> First plan i use >>> session.execute("bridge", >>> "[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/ >>> xxxx >>> @provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/ >>> exte >>> rnal/xxxx at provder2"); i modify nibblebill for match provider with my >>> billing. >>> this case still fail. >>> >>> now i try >>> >>> if >>> (session.ready()){ >>> s = new >>> Session("{absolute_codec_string='GSM,G729'}sofia/external/ >>> xxx at provider1" >>> >>> } >>> if >>> (s.ready()){ >>> session.execute("set", >>> "nibble_rate=2.5"); >>> session.execute("set", >>> "nibble_account="+acaller); >>> session.execute("set", >>> "hangup_after_bridge=false"); >>> session.execute("set", >>> "provider_id="+dialprovider_id[1]); >>> >>> bridge(session,s); >>> } >>> >>> and check hangup cause before try other provider. >>> >>> >>> >>> Please guide me it's right way or not ? >>> >>> >>> Dome C. >>> >>> >>> 2009/6/26 Darren Schreiber >>>> >>>> Did this work? Would love an update on this error/issue. >>>> ________________________________ >>>> From: Michael Jerris [mailto:mike at jerris.com] >>>> Sent: Wednesday, June 24, 2009 8:15 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway >>>> >>>> try adding >>>> before the >>>> bridge and report back results. >>>> Mike >>>> On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote: >>>> >>>> Dear All, >>>> >>>> Look like nibblebill does't work with multiple gatreway. >>>> I try >>>> >>> data="nibble_account=0838833133"/> >>>> >>> data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/ >>>> extern >>>> al/6626734000 at 203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/ >>>> 6626734 >>>> 000 at 202.xxx.xxx.xxx> >>>> >>>> nibblebill not found nibble_rate >>>> >>>> But >>>> >>>> >>> data="nibble_account=0838833133"/> >>>> >>> data="{absolute_codec_string='GSM,G729'}sofia/external/ >>>> 6626734000 at 203 >>>> .xxx.xxx.xxx|sofia/external/6626734000 at 202.xxx.xxx.xxx> >>>> >>>> Work fine >>>> >>>> What's difference from set application and [] ? >>>> >>>> Best Regards. >>>> Dome C. >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>> ers >>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>>> ers >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> rs >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> rs >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mail at willboyce.com Fri Jun 26 08:43:27 2009 From: mail at willboyce.com (Will Boyce) Date: Fri, 26 Jun 2009 10:43:27 -0500 (CDT) Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <10974959.701246030807467.JavaMail.SYSTEM@man-00108> Message-ID: <8044499.721246030958844.JavaMail.SYSTEM@man-00108> Hey David, >> There doesn't seem to be any direct link between FreeSWITCH and >> Howler. >> How does this benefit the project? Can we assume that Howler is >> "giving >> something back"? > > Well, they're providing additional functionality which a lot of folk > want, which requires substantial investment in money (visit sipro.com > and have a look at the up-front license fees if you wish) and time (to > get the thing working across the various platforms with the required > licensing) and they've presumably taken a commercial risk so to do. > > Hats off to them: they've taken a risk, and the project benefits > immensely from it. Expecting them to "give something back" in > addition > would be entirely unreasonable. You're spot on on the somewhat extortionate up-front (and continuing royalty) fees, and there is substantial risk in us doing this which we hope will be mitigated by the rising interest in FreeSWITCH as a whole (a project which we've been huge evangelists of since the beginning). We hope that adding patent encumbered codecs like G.729 to FreeSWITCH will ultimately benefit the project and cause more people to be able to transition towards it from other platforms, building on the fantastic platform that Anthony and the rest of the gang have somehow managed to develop for free! Our initial feedback has been very promising, with a number of customers at least in the UK (including one major VoIP provider) that we've spoken to directly now seriously considering a migration to FreeSWITCH. We hope that the small per channel charge is not too onerous - we've tried to make it as cheap as possible by introducing the floating licenses (and including all support and maintenance upgrades free of charge), but there are very real costs here in royalties and development - how you guys do it mostly for free is beyond me! We have, of course, spoken to the FreeSWITCH team and offered to collaborate on this project (as they've also been working on this for a while) and we hope that we can join forces on it soon so that everyone benefits :) -- Regards, Jay Fenton, CTO c/o Will Boyce Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton From mike at jerris.com Fri Jun 26 09:03:10 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 12:03:10 -0400 Subject: [Freeswitch-users] From Asterisk to Freeswitch In-Reply-To: <8ccbff060906252345w1aa7c4a1u66f1c2af752dea20@mail.gmail.com> References: <8ccbff060906252345w1aa7c4a1u66f1c2af752dea20@mail.gmail.com> Message-ID: <8C494316-C254-42F5-BF69-ED687AF2E6E0@jerris.com> most of the information about fifo is : http://wiki.freeswitch.org/wiki/Mod_fifo Mike On Jun 26, 2009, at 2:45 AM, Dome Charoenyost wrote: > Dear All, > > I'm asterisk developer(I have some code in Asterisk) . After 3 > weeks with freeswich nothing to say. now i'm move all callingcard , > wholesale platfrom to Freeswitch. I'm very happy with bridge , > nibblebill. after finish this job i'll test FS PBX feature. i think > it's easy to do Hosted IPPBX. But i want to know more about mod_fifo > Can someone tell me about mod_fifo compare with asterisk app_queue. > i'm talking about annouce , priority agent > From mike at jerris.com Fri Jun 26 09:05:52 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 12:05:52 -0400 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> Message-ID: <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> I would pre-adjust the volume of the soundfile with sox instead of doing it real time. Mike On Jun 26, 2009, at 7:38 AM, Dennis wrote: > hi, > > we want to use uuid_displace with mux to playback a soundfile to a > bridged uuid, so that this uuid can hear the other side talk AND hear > the soundfile (whispering). > > is there an option we can set, for defining the loudness level of the > soundfile? in our tests the soundfile was way to loud, so that it was > nearly impossible to hear the other side talk, while the soundfile was > playing. > > we tried "uuid_displace uuid start /path/to/soundfile/soundfile.wav 0 > mux 0.3", so that the loudness of soundfile only would be 30% - but > this does not work. > From solko at gcdf.pl Fri Jun 26 09:14:10 2009 From: solko at gcdf.pl (Szymon Olko) Date: Fri, 26 Jun 2009 18:14:10 +0200 Subject: [Freeswitch-users] Bug reports In-Reply-To: References: Message-ID: <4A44F3D2.10901@gcdf.pl> Brian West pisze: > FreeSWITCHers, > > We have written an extensive guide on posting bugs to jira. Over the > past few weeks everyone has been a little lax in posting the correct > info the first time. The questions we ask are NOT optional to fill > out.. You must fill out every question on the list and "doesn't apply" > or "n/a" are not acceptable answers. So help us help you fix your > issues that you report. Remember provide all the details even if you > think its not relevant to the issue at hand because even the smallest > detail can help me reproduce your issue or result in me pulling my hair > out and going crazy. :P > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Also I would like to solicit volunteers to help manage, track and > collect info for any Jira's please contact me... I have asked a few > times but very few expressed interest in helping. > I agree that those information are needed for developer, I will fill them all. What would make it easy for me if there was a "clone" option to make new ticked based on old one but with ability to change information. Most of things you ask are the same in all my tickes. I would just need to put new bug description and change only few items. If this is not possible then maybe jira has something like "ticket template" which I can fill once and then make all tickets base on it. Regards Szymon PS: Wiki page is great > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Fri Jun 26 09:37:22 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 26 Jun 2009 19:37:22 +0300 Subject: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH In-Reply-To: <8044499.721246030958844.JavaMail.SYSTEM@man-00108> References: <8044499.721246030958844.JavaMail.SYSTEM@man-00108> Message-ID: <1246034242.4232.25.camel@dk-d820> On Fri, 2009-06-26 at 10:43 -0500, Will Boyce wrote: > Hey David, [duplicate post snipped] -- don't suppose your product includes an echo canceller, does it ;-) Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From shiyanov at gmail.com Fri Jun 26 10:25:50 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Fri, 26 Jun 2009 21:25:50 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge Message-ID: Hello! I got a problem with one way audio, symptoms are: firstly play audio file to channel A (A is hears sound) secondly bridge channel B with A (A doesn't hear B). Environment: - no NAT - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch - dialplan: - Call routing scheme: user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc Exact description what's going on is: user A -> FS -(bridge)-> my B2BUA Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) user to extension "Local_Extension_from_SP". This should create a new call to user B. As a result - A doesn't hear B, but B- is OK. On the contrary, if a call is routed (by B2BUA) to the "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - everything is OK. What I've tried: - set parameter "inbound-proxy-media" to "true" in Sofia profile - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile Nothing helps. Any help or thoughts would be MUCH appreciated! Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/7e19e993/attachment-0002.html From msc at freeswitch.org Fri Jun 26 10:46:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 10:46:34 -0700 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> Message-ID: <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> On Fri, Jun 26, 2009 at 9:05 AM, Michael Jerris wrote: > I would pre-adjust the volume of the soundfile with sox instead of > doing it real time. > > Mike > I have to agree with Mike here. Sox is awesome for this kind of thing and disk space is way more plentiful than CPU power. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/a51a2cd1/attachment-0002.html From msc at freeswitch.org Fri Jun 26 10:50:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 10:50:10 -0700 Subject: [Freeswitch-users] Service_not_implemented for mobile phones? In-Reply-To: <24217116.post@talk.nabble.com> References: <24217116.post@talk.nabble.com> Message-ID: <87f2f3b90906261050q6ff09ac2ka0a6e06f3c8a924d@mail.gmail.com> Edmar, I strongly recommend you review the troubleshooting page on the wiki. No one will be able to help you without more information about your issue. http://wiki.freeswitch.org/wiki/Reporting_Bugs Be sure to use pastebin and capture the debug output from the FS command line and also paste your relevant configuration files. -MC On Fri, Jun 26, 2009 at 2:03 AM, Edmar Cruz wrote: > > Hi, > > I receive an error message service not implemented sometimes when calling > a mobile phone number but sometimes it works. What maybe rhe cause of this > error? I already installed zfone, perfectly connect to two freeswitch and > the one issue I got today is these can you help me guys? > > > Thanks > -- > View this message in context: > http://www.nabble.com/Service_not_implemented-for-mobile-phones--tp24217116p24217116.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/d5aa1942/attachment-0002.html From brian at freeswitch.org Fri Jun 26 10:52:52 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 12:52:52 -0500 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> Message-ID: <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> Are you playing a stereo file by chance? /b On Jun 26, 2009, at 12:46 PM, Michael Collins wrote: > > > On Fri, Jun 26, 2009 at 9:05 AM, Michael Jerris > wrote: > I would pre-adjust the volume of the soundfile with sox instead of > doing it real time. > > Mike > > I have to agree with Mike here. Sox is awesome for this kind of > thing and disk space is way more plentiful than CPU power. > > -MC > _______ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/98b6efed/attachment-0002.html From brian at freeswitch.org Fri Jun 26 10:54:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 12:54:00 -0500 Subject: [Freeswitch-users] Bug reports In-Reply-To: <4A44F3D2.10901@gcdf.pl> References: <4A44F3D2.10901@gcdf.pl> Message-ID: <58A2C686-BEE7-4F17-825C-FCFE922F07F5@freeswitch.org> Well I'm going to try my best to reproduce issues on demand but sometimes I just can't, other times the reporter stops responding and I have no choice but to close the issue out. :) /b On Jun 26, 2009, at 11:14 AM, Szymon Olko wrote: > I agree that those information are needed for developer, I will fill > them all. > > What would make it easy for me if there was a "clone" option to make > new ticked based on old one but with ability to change > information. Most of things you ask are the same in all my tickes. I > would just need to put new bug description and change only > few items. > If this is not possible then maybe jira has something like "ticket > template" which I can fill once and then make all tickets base > on it. > > Regards > Szymon > > PS: Wiki page is great From brian at freeswitch.org Fri Jun 26 10:55:42 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 12:55:42 -0500 Subject: [Freeswitch-users] Bug reports In-Reply-To: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> References: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> Message-ID: It involves more than just reporting... it involves trying to reproduce the issue or clarify the steps to reproduce an issue.. I have been thru some issues that sound crazy hard to reproduce... like stand on your head .. dial 1234, spin around dial 456, scream into the phone... dance a little jig... then hangup... I know it sound silly but there have been bugs that are so complex to reproduce you end up pulling your hair out :P Granted they are few and far between and mostly corner cases. /b On Jun 26, 2009, at 10:26 AM, Chris Chen wrote: > Brian, I would like to be one of the volunteers helping to report > issues. > > Chris From shiyanov at gmail.com Fri Jun 26 10:59:38 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Fri, 26 Jun 2009 21:59:38 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: Updates: 1. One-way audio is in 95% tries. But how the rest 5% works?? 2. Strange FS logging after the channels are bridged (user A talk to user B) 2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at 192.168.147.1 entering state [ready] 2009-06-26 02:16:07 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 192.168.147.130 s=FreeSWITCH c=IN IP4 192.168.147.130 t=0 0 m=audio 31134 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 0 RTP/AVP 34 a=rtpmap:34 H263/90000 2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1000000000 at 192.168.147.130:5060 entering state [ready] freeswitch at localhost.localdomain> 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/ 1005 at uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:17:09 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 freeswitch at localhost.localdomain> show calls API CALL [show(calls)] output: created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2009-06-26 02:16:05,1245968165,switch_ivr_multi_threaded_bridge,1005,1005,inbound1000000000,sofia/external/ 1005 at uat.pbx.starpoundtech.net ,4fa86434-b542-4066-99af-5924c78ddab7,1005,1005, 1000000000 at 192.168.147.130:5060,sofia/external/ 1000000000 at 192.168.147.130:5060,73df8735-fee2-464d-aec0-fda886ba2cba 2009-06-26 02:16:07,1245968167,switch_ivr_multi_threaded_bridge,1005,1005,1001,sofia/external/ 1005 at 192.168.147.1 ,1c2c5f6d-669f-4432-ad04-35a64dbc8a14,1005,1005,sip:1001 at 192.168.147.1:5060 ;fs_nat=yes,sofia/doublenat5090/sip:1001 at 192.168.147.1:5060 ;fs_nat=yes,66895f68-70bf-410a-bff7-cda9549c102d 2 total. freeswitch at localhost.localdomain> 2009-06-26 02:18:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/ 1005 at uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:18:10 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:18:10 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-06-26 02:19:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [calling] 2009-06-26 02:19:08 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready] 2009-06-26 02:19:08 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1 3 IN IP4 192.168.147.1 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.147.1 t=0 0 m=audio 47590 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Artem On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov wrote: > Hello! > > I got a problem with one way audio, symptoms are: > firstly play audio file to channel A (A is hears sound) > secondly bridge channel B with A (A doesn't hear B). > > Environment: > - no NAT > - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of > them- no audio, Wireshark shows that there is no RTP-flow to A from > FreeSwitch > - dialplan: > > > > > > > > > > > > > > > > expression="^${caller_id_number}$"> > > data="transfer_ringback=${us-ring}"/> > > > data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > data="sip_h_X-SPFrom="e;${sip_from_user}"e;<${sip_from_uri}>"/> > data="sip_h_X-SPTo=<${sip_to_uri}>"/> > data="sip_h_X-SPCallId=${sip_call_id}"/> > data="sofia/external/${orgname}send2voicemail@ > $${starpound_sip_app_server}"/> > > > - Call routing scheme: > user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc > Exact description what's going on is: > user A -> FS -(bridge)-> my B2BUA > Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to > extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) > user to extension "Local_Extension_from_SP". This should create a new call > to user B. As a result - A doesn't hear B, but B- is OK. > On the contrary, if a call is routed (by B2BUA) to the > "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - > everything is OK. > > > What I've tried: > - set parameter "inbound-proxy-media" to "true" in Sofia profile > - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile > Nothing helps. > > > Any help or thoughts would be MUCH appreciated! > Artem > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/f49f68f1/attachment-0002.html From brian at freeswitch.org Fri Jun 26 11:03:40 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 13:03:40 -0500 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk also... due to the lines below. /b On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 > 192.168.147.130 > s=FreeSWITCH From Mailings at kh-dev.de Fri Jun 26 12:48:36 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 26 Jun 2009 21:48:36 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Message-ID: Hi all, I'm just writing a perl script as dialplan to learn how to handle freeswitch. Now I have the issue that my hangup hook won't be triggered after I intercepted a call. A "normal" hangup triggers the function. Does anybody have a hint how to get the hangup hook triggered? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/503b4419/attachment-0002.html From brian at freeswitch.org Fri Jun 26 13:06:52 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 15:06:52 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: Message-ID: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> Depends what side of the call was the hangup hook on? /b On Jun 26, 2009, at 2:48 PM, Klaus Hochlehnert wrote: > Hi all, > > I?m just writing a perl script as dialplan to learn how to handle > freeswitch. > > Now I have the issue that my hangup hook won?t be triggered after I > intercepted a call. > A ?normal? hangup triggers the function. > > Does anybody have a hint how to get the hangup hook triggered? > > Thanks, Klaus > ______________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/ece4df34/attachment-0002.html From Mailings at kh-dev.de Fri Jun 26 13:22:14 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 26 Jun 2009 22:22:14 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> Message-ID: Actually one of my first actions in the script is $session->setHangupHook('on_hangup'); When a call comes in the hook is set and working. The second time the script is called when I try to intercept. As it's the same script there's also the function setHangupHook called. That's what I've currently done. How can I set up the hook for the "new" bridge? Or is there a possibility to set a global hook? Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 26, 2009 10:07 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Depends what side of the call was the hangup hook on? /b On Jun 26, 2009, at 2:48 PM, Klaus Hochlehnert wrote: Hi all, I'm just writing a perl script as dialplan to learn how to handle freeswitch. Now I have the issue that my hangup hook won't be triggered after I intercepted a call. A "normal" hangup triggers the function. Does anybody have a hint how to get the hangup hook triggered? Thanks, Klaus ______________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/c326f510/attachment-0002.html From brian at freeswitch.org Fri Jun 26 13:27:32 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 15:27:32 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> Message-ID: <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> well in your case I suspect your intercepting the leg of the call without the hook on it. /b On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: > Actually one of my first actions in the script is > $session->setHangupHook('on_hangup'); > > When a call comes in the hook is set and working. > > The second time the script is called when I try to intercept. As > it?s the same script there?s also the function setHangupHook called. > That?s what I?ve currently done. > > How can I set up the hook for the ?new? bridge? > Or is there a possibility to set a global hook? > > Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/ba8c9d6c/attachment-0002.html From Mailings at kh-dev.de Fri Jun 26 13:39:16 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Fri, 26 Jun 2009 22:39:16 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> Message-ID: Ok, and how can I ask the hook to come with me? If I understand this right... When a call comes in the hook is set on the a-leg and it rings on the b-leg. When I do an intercept I kill the ringing b-leg and the interceptor is now the "new" b-leg, right? I would assume that the "old" a-leg still has the hook on it or this wrong. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 26, 2009 10:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... well in your case I suspect your intercepting the leg of the call without the hook on it. /b On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: Actually one of my first actions in the script is $session->setHangupHook('on_hangup'); When a call comes in the hook is set and working. The second time the script is called when I try to intercept. As it's the same script there's also the function setHangupHook called. That's what I've currently done. How can I set up the hook for the "new" bridge? Or is there a possibility to set a global hook? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/1cee6d64/attachment-0002.html From msc at freeswitch.org Fri Jun 26 13:56:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 13:56:43 -0700 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> Message-ID: <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> Can you paste in your script so we can see what is going on? -MC On Fri, Jun 26, 2009 at 1:39 PM, Klaus Hochlehnert wrote: > Ok, and how can I ask the hook to come with me? > > > > If I understand this right... > > When a call comes in the hook is set on the a-leg and it rings on the > b-leg. > > When I do an intercept I kill the ringing b-leg and the interceptor is now > the ?new? b-leg, right? > > I would assume that the ?old? a-leg still has the hook on it or this wrong. > > > > Thanks, Klaus > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Friday, June 26, 2009 10:28 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] hangup hook after intercept doesn't get > triggered... > > > > well in your case I suspect your intercepting the leg of the call without > the hook on it. > > > > /b > > > > On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: > > > > Actually one of my first actions in the script is > > $session->setHangupHook('on_hangup'); > > > > When a call comes in the hook is set and working. > > > > The second time the script is called when I try to intercept. As it?s the > same script there?s also the function setHangupHook called. > > That?s what I?ve currently done. > > > > How can I set up the hook for the ?new? bridge? > > Or is there a possibility to set a global hook? > > > > Thanks, Klaus > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/5da26bc9/attachment-0002.html From john at feith.com Fri Jun 26 14:33:05 2009 From: john at feith.com (John Wehle) Date: Fri, 26 Jun 2009 17:33:05 -0400 (EDT) Subject: [Freeswitch-users] Accessing a global variable from lua Message-ID: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> How do you get a system variable from within a lua startup script? Specifically I want domain_name from vars.xml ... normally I'd use session:getVariable, however there is no session in this case. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From brian at freeswitch.org Fri Jun 26 14:37:48 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 16:37:48 -0500 Subject: [Freeswitch-users] Accessing a global variable from lua In-Reply-To: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> References: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> Message-ID: <60AC4243-2C80-4A0D-8B7E-F908D49A5BA7@freeswitch.org> You can execute global_getvar api call. /b On Jun 26, 2009, at 4:33 PM, John Wehle wrote: > How do you get a system variable from within a lua startup script? > Specifically I want domain_name from vars.xml ... normally I'd use > session:getVariable, however there is no session in this case. From msc at freeswitch.org Fri Jun 26 14:41:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 26 Jun 2009 14:41:50 -0700 Subject: [Freeswitch-users] Accessing a global variable from lua In-Reply-To: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> References: <200906262133.n5QLX5hU005468@jwlab.FEITH.COM> Message-ID: <87f2f3b90906261441oe79997fod20c0ddb291d3984@mail.gmail.com> Use the API to execute global_getvar... http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls -MC On Fri, Jun 26, 2009 at 2:33 PM, John Wehle wrote: > How do you get a system variable from within a lua startup script? > Specifically I want domain_name from vars.xml ... normally I'd use > session:getVariable, however there is no session in this case. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | > | John Wehle | Fax: 1-215-540-5495 | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/41c75cfa/attachment-0002.html From john at feith.com Fri Jun 26 15:13:12 2009 From: john at feith.com (John Wehle) Date: Fri, 26 Jun 2009 18:13:12 -0400 (EDT) Subject: [Freeswitch-users] Accessing a global variable from lua Message-ID: <200906262213.n5QMDCgh005662@jwlab.FEITH.COM> > You can execute global_getvar api call. Thanks ... I've updated the wiki. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john at feith.com | | John Wehle | Fax: 1-215-540-5495 | | ------------------------------------------------------------------------- From brian at freeswitch.org Fri Jun 26 15:31:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 17:31:19 -0500 Subject: [Freeswitch-users] Accessing a global variable from lua In-Reply-To: <200906262213.n5QMDCgh005662@jwlab.FEITH.COM> References: <200906262213.n5QMDCgh005662@jwlab.FEITH.COM> Message-ID: John, Also can you go over the few jira's you have left and see if we can resolve/knock out some of them.. I'm wanting to roll pre9 this weekend. Thanks, Brian On Jun 26, 2009, at 5:13 PM, John Wehle wrote: >> You can execute global_getvar api call. > > Thanks ... I've updated the wiki. > > -- John > ------------------------------------------------------------------------- > | Feith Systems | Voice: 1-215-646-8000 | Email: > john at feith.com | > | John Wehle | Fax: 1-215-540-5495 > | | > ------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Mailings at kh-dev.de Fri Jun 26 16:16:31 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sat, 27 Jun 2009 01:16:31 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> Message-ID: Brian told me to open a jira, what I did now. But here's the script. It basically writes incoming calls to a database and removes them after hangup. When intercepting it also rewrites the destination of the call. Thanks, Klaus #!/usr/bin/perl use strict; use DBI; use POSIX qw(strftime); our $session; use constant { false => 0, true => 1, }; my $EmptyString = " "; # Used to delete regexp catches my $dbargs = {AutoCommit => 0, PrintError => 1}; my $dbh = DBI->connect("dbi:SQLite:dbname=/opt/freeswitch/db/dialplan_call_info.db", "", "", $dbargs); $session->setHangupHook('on_hangup'); logInfo("Hook set"); sub logString { my ($Level, $Msg) = @_; freeswitch::consoleLog("$Level", "$Msg\n"); } sub logDebug { my ($Msg) = @_; logString("DEBUG", ">>>>> $Msg"); } sub logInfo { my ($Msg) = @_; logString("INFO", ">>>>> $Msg"); } sub logNotice { my ($Msg) = @_; logString("NOTICE", ">>>>> $Msg"); } sub logWarning { my ($Msg) = @_; logString("WARNING", ">>>>> $Msg"); } sub logError { my ($Msg) = @_; logString("ERR", ">>>>> $Msg"); } sub logCritical { my ($Msg) = @_; logString("CRIT", ">>>>> $Msg"); } sub logAlert { my ($Msg) = @_; logString("ALERT", ">>>>> $Msg"); } # The idea of these functions is to allow for easy pull in of variables and then # automatically export any ones that have been changed when UPDATEV. # It will ensure you don't write to any non-imported variables, but as we are # using a hash we cannot prevent invalid reads. If you are really concerned about # this then you could use a specific read function which first checks to make sure # its defined in CLEAN_VARS before returning. my %VARS; my %CLEAN_VARS; # Takes one or more variables names to import in sub GETV { my @Arr = @_; foreach my $Var (@Arr) { $VARS{$Var} = $session->getVariable("$Var"); $CLEAN_VARS{$Var} = $VARS{$Var}; if (! defined $CLEAN_VARS{$Var}) { $CLEAN_VARS{$Var} = ""; } } } # Generally not called directly, but will set the variable to the value requested right away sub SETV { my ($Var, $Value) = @_; $session->setVariable("$Var", "$Value"); $VARS{$Var} = "$Value"; $CLEAN_VARS{$Var} = "$Value"; } # If we don't care about a variables value, but wan't to override it this will add it to the hash # so that when we write to it, we don't consider it a typo sub ADDV { my @Arr = @_; foreach my $Var(@Arr) { $CLEAN_VARS{$Var} = "123zzzzzZnzZZzz"; # Something definitely won't match $VARS{$Var} = ""; } } # Updates any changed variables sub UPDATEV { foreach my $Var (keys %VARS) { # Make sure there were no typos if (! defined $CLEAN_VARS{$Var}) { die "Warning a variable of: '$Var' was not found in CLEAN_VARS, did you forget to GET/ADD it first?"; } if ($VARS{$Var} ne $CLEAN_VARS{$Var}) { SETV($Var, $VARS{$Var}); } } } # Dump all variables sub DUMPV { foreach my $Var (sort keys %VARS) { logInfo("$Var = '" . $VARS{$Var} . "'"); } } sub CAN_ACCESS { my ($Req) = @_; if ($VARS{app_rights} eq "ALL" || $VARS{app_rights} =~ /#$Req#/) { return true; } else { return false; } } # Fetch some generic variables GETV("uuid", "base_dir", "domain", "app_rights", "de-ring", "outgoing_soundtouch_profile", "hold_music", "continue_on_fail"); GETV("destination_number", "caller_id_name", "caller_id_number", "effective_caller_id_name", "effective_caller_id_number"); GETV("network_addr", "hangup_after_bridge", "called_party_callgroup", "ringback", "transfer_ringback", "sip_exclude_contact"); GETV("call_timeout", "source", "sip_to_params", "presence_id", "dialed_user", "dialed_domain"); GETV("voicemail_authorized", "sip_authorized", "username", "accountcode", "sip_from_user", "sip_to_user"); # Set some defaults $VARS{hangup_after_bridge} = "true"; $VARS{ringback} = $VARS{'de-ring'}; $VARS{transfer_ringback} = $VARS{hold_music}; $VARS{sip_exclude_contact} = $VARS{network_addr}; $VARS{call_timeout} = "60"; $VARS{continue_on_fail} = "true"; # NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION UPDATEV(); sub bridgeCallInternally { my ($DestNr) = @_; if ("${DestNr}" == "21") { $VARS{call_timeout} = "15"; } UPDATEV(); $dbh->do("insert into current_calls (extension, uuid) values ('$DestNr', '$VARS{uuid}')"); $dbh->commit(); $session->execute("record_session","\${base_dir}/recordings/\${strftime(%Y-%m-%d-%H-%M-%S)}_\${destination_number}_\${caller_id_number}.wav"); $session->execute("bridge","user/${DestNr}\@$VARS{domain}"); } sub on_hangup { my $hup_session = shift; my $hup_cause = shift; logInfo("Hangup uuid: '" . $hup_session->{uuid} . "'"); logInfo("Hangup cause: '$hup_cause'"); $dbh->do("delete from current_calls where uuid = '" . $hup_session->{uuid} . "'"); $dbh->commit(); } # Internal numbers if ($VARS{destination_number} =~ /^(2[0-2])$/) { UPDATEV(); bridgeCallInternally($VARS{destination_number}); } # Intercept call if ($VARS{destination_number} =~ /^\*8(\d+)$/) { my $intercept_extension = ""; my $intercept_uuid = ""; my $sth = $dbh->prepare("select * from current_calls where extension = ?"); $sth->execute($1); while (my @data = $sth->fetchrow_array()) { $intercept_extension = $data[0]; $intercept_uuid = $data[1]; } logInfo("Intercept call from '$intercept_extension' - '$intercept_uuid'"); GETV("caller_id_number"); $dbh->do("update current_calls set extension = '$VARS{caller_id_number}' where uuid = '$intercept_uuid'"); $dbh->commit(); $session->answer(); $session->execute("intercept", "$intercept_uuid"); $session->execute("sleep", "1000"); } $dbh->disconnect(); return 1; From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, June 26, 2009 10:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Can you paste in your script so we can see what is going on? -MC On Fri, Jun 26, 2009 at 1:39 PM, Klaus Hochlehnert > wrote: Ok, and how can I ask the hook to come with me? If I understand this right... When a call comes in the hook is set on the a-leg and it rings on the b-leg. When I do an intercept I kill the ringing b-leg and the interceptor is now the "new" b-leg, right? I would assume that the "old" a-leg still has the hook on it or this wrong. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 26, 2009 10:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... well in your case I suspect your intercepting the leg of the call without the hook on it. /b On Jun 26, 2009, at 3:22 PM, Klaus Hochlehnert wrote: Actually one of my first actions in the script is $session->setHangupHook('on_hangup'); When a call comes in the hook is set and working. The second time the script is called when I try to intercept. As it's the same script there's also the function setHangupHook called. That's what I've currently done. How can I set up the hook for the "new" bridge? Or is there a possibility to set a global hook? Thanks, Klaus _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/32ccd95d/attachment-0002.html From brian at freeswitch.org Fri Jun 26 16:27:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 18:27:47 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> Message-ID: <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> Please open the jira and attach all this and the xml dialplan to execute it .. also the schema for the db would be helpful also. Thanks, /b On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote: > Brian told me to open a jira, what I did now. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/962bcd6b/attachment-0002.html From Mailings at kh-dev.de Fri Jun 26 16:36:34 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sat, 27 Jun 2009 01:36:34 +0200 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> Message-ID: I used the Project "FS Scripts". Hope that's ok. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, June 27, 2009 1:28 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't get triggered... Please open the jira and attach all this and the xml dialplan to execute it .. also the schema for the db would be helpful also. Thanks, /b On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote: Brian told me to open a jira, what I did now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/aabf0f56/attachment-0002.html From brian at freeswitch.org Fri Jun 26 16:40:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 18:40:38 -0500 Subject: [Freeswitch-users] hangup hook after intercept doesn't get triggered... In-Reply-To: References: <7451D164-A58F-4DB5-8226-622D9C5A762D@freeswitch.org> <7B59EA3E-D3DE-446C-B550-E0DE268AB88B@freeswitch.org> <87f2f3b90906261356y774cb7a0u75bc29ec0fc3b71f@mail.gmail.com> <19F89597-0838-41AB-820F-249B7759C201@freeswitch.org> Message-ID: <16FDA78B-5681-410D-AECA-5C50C9FD4A73@freeswitch.org> Its more FSCORE but i'll move it into the right place ;) /b On Jun 26, 2009, at 6:36 PM, Klaus Hochlehnert wrote: > I used the Project ?FS Scripts?. Hope that?s ok. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Saturday, June 27, 2009 1:28 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] hangup hook after intercept doesn't > get triggered... > > Please open the jira and attach all this and the xml dialplan to > execute it .. also the schema for the db would be helpful also. > > Thanks, > /b > > On Jun 26, 2009, at 6:16 PM, Klaus Hochlehnert wrote: > > > Brian told me to open a jira, what I did now. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/f2529776/attachment-0002.html From jason at jasonjgw.net Fri Jun 26 17:02:14 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 10:02:14 +1000 Subject: [Freeswitch-users] Bug reports In-Reply-To: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> References: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> Message-ID: <20090627000214.GA7509@jdc.jasonjgw.net> Chris Chen wrote: > Brian, I would like to be one of the volunteers helping to report issues. That's great. We need more volunteers. For some FreeSWITCH users (of whom I am one), the user interface of Jira is an obstacle to reporting bugs via that mechanism, for accessibility reasons. I've had better luck with Bugzilla and other bug tracking systems, but Jira really seems to depend on Javascript for its fundamental operations, and that's a real barrier for me at the moment. From jason at jasonjgw.net Fri Jun 26 17:05:41 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 10:05:41 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <4A44AB08.1050503@gmail.com> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> Message-ID: <20090627000541.GB7509@jdc.jasonjgw.net> Tamas wrote: > Did you make bootstrap.sh and configure before compilation? Yes. This was a clean export from svn, built by running the Debian debuild tool, as in svn export to a temporary directory, followed by debuild (after changing the version number to make the package version unique). From brian at freeswitch.org Fri Jun 26 17:10:35 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 19:10:35 -0500 Subject: [Freeswitch-users] Bug reports In-Reply-To: <20090627000214.GA7509@jdc.jasonjgw.net> References: <507898380906260826i31dcc405h6dd4757588dfd79a@mail.gmail.com> <20090627000214.GA7509@jdc.jasonjgw.net> Message-ID: Jason, In your case we'll gladly accept your bug reports via the mailing list for that exact reason. ;) /b On Fri, Jun 26, 2009 at 7:02 PM, Jason White wrote: > Chris Chen wrote: > > Brian, I would like to be one of the volunteers helping to report issues. > > That's great. We need more volunteers. > > For some FreeSWITCH users (of whom I am one), the user interface of Jira is > an > obstacle to reporting bugs via that mechanism, for accessibility reasons. > I've > had better luck with Bugzilla and other bug tracking systems, but Jira > really > seems to depend on Javascript for its fundamental operations, and that's a > real barrier for me at the moment. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/3824cae4/attachment-0002.html From brian at freeswitch.org Fri Jun 26 17:10:58 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 19:10:58 -0500 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090627000541.GB7509@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> Message-ID: what are the error messages? /b On Fri, Jun 26, 2009 at 7:05 PM, Jason White wrote: > Tamas wrote: > > Did you make bootstrap.sh and configure before compilation? > > Yes. This was a clean export from svn, built by running the Debian debuild > tool, as in > > svn export to a temporary directory, followed by debuild (after changing > the > version number to make the package version unique). > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/6b7326f1/attachment-0002.html From jason at jasonjgw.net Fri Jun 26 17:31:40 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 10:31:40 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> Message-ID: <20090627003140.GA12881@jdc.jasonjgw.net> Brian West wrote: > what are the error messages? There aren't any. The build completes without error, but the module doesn't load due to the undefined symbols. From wiltingtree at gmail.com Fri Jun 26 19:10:10 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Fri, 26 Jun 2009 22:10:10 -0400 Subject: [Freeswitch-users] Version 1.0.4 not working with custom channel variables? Message-ID: Hi, I recently tried upgrading FreeSWITCH from 1.0.2 to 1.0.4pre8 on my server, but the application I wrote uses custom channel variables which I create using "setVariable" in Python. In version 1.0.2, I am able to retrieve that variable within the event xml (it prepends the variable name with variable_). But in version 1.0.4pre8, these variables are missing from the xml events. Could this be a bug? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/824b2ffc/attachment-0002.html From vince.freeswitch at hightek.org Fri Jun 26 19:17:44 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Fri, 26 Jun 2009 21:17:44 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> Message-ID: <20090627021744.GA89233@quark.hightek.org> On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > > Can you post a bug to Jira.freeswitch.org with all these warnings, > even better with patches to fix it. OK. I think I have narrowed the problem down to 3 issues. 1. The build system is treating even a single warning as a critical error and aborting compilation in that directory. 2. Compilation continues into the next directory even though compilation of the previous directory was aborted, as you can see in the make output below where it went on to build "features". This can cause a chain reaction of other errors because stuff it expects to be there did not get built in previous stages. I suspect that is also the source of the 'symbolic link' error I was getting. I did not get that on this last compilation after fixing some of the warnings (see below). It can also make it possible to get to the end of the build and not know that stuff did not get compiled further back, leaving the package incomplete. 3. Lots of "return makes pointer from integer without a cast" warnings throughout the sofia-sip tree. This one, of course, is not the actual show stopper but is triggering the above problems and needs cleaned up none the less. ================= Making all in su LTCOMPILE su.lo LTCOMPILE su_errno.lo LTCOMPILE su_addrinfo.lo LTCOMPILE su_alloc.lo su_alloc.c: In function `sub_alloc': su_alloc.c:428: warning: return makes pointer from integer without a cast su_alloc.c:511: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_home_new': su_alloc.c:555: warning: return makes pointer from integer without a cast su_alloc.c:557: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_home_clone': su_alloc.c:730: warning: return makes pointer from integer without a cast su_alloc.c:732: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_realloc': su_alloc.c:1315: warning: return makes pointer from integer without a cast su_alloc.c:1319: warning: return makes pointer from integer without a cast su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features ... ================= I confirmed the abort on warning issue (1) by fixing all warnings in su_alloc.c. As you can see below, it went past it just fine until it got a warning in su_sprintf.c. ================= Making all in su LTCOMPILE su.lo LTCOMPILE su_errno.lo LTCOMPILE su_addrinfo.lo LTCOMPILE su_alloc.lo LTCOMPILE su_alloc_lock.lo LTCOMPILE su_strdup.lo LTCOMPILE su_sprintf.lo su_sprintf.c: In function `su_vsprintf': su_sprintf.c:98: warning: return makes pointer from integer without a cast gmake[9]: *** [su_sprintf.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features ... ================= The issue below of saying it was successfull, when it was not, is apparently part of issue 2. It continued into the "build" directory after the previous errors. ================= Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 ================= I went ahead and established an account on jira.freeswitch.org. I see there are separate projects for "FreeSWITCH-Buildsystem" and "sofia-sip". I am guessing I should post two separate reports, one for issues 1 and 2 for the Buildsystem and one in sofia-sip for issue 3 since all the warnings seem to be from that code. I do not know the build system well enough to solve issues 1 and 2 at this time. However, I already have a patch I will provide for su_alloc.c and can continue working on resolving more of the casting warnings if somebody else can work on the build issue, which is the real show stopper. From vince.freeswitch at hightek.org Fri Jun 26 19:19:50 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Fri, 26 Jun 2009 21:19:50 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <9E2119E8-99A3-40B9-960F-539B881CF1EE@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <20090624031950.GD2623@hijacked.us> <20090625214910.GB45220@quark.hightek.org> <9E2119E8-99A3-40B9-960F-539B881CF1EE@jerris.com> Message-ID: <20090627021950.GB89233@quark.hightek.org> On Thu, Jun 25, 2009 at 06:06:04PM -0400, Michael Jerris wrote: > > > Is there a log file somewhere that contains the actual compile > > commands and error output so you can find out what happened when > > there is a error? Or perhaps a configuration to enable it to come > > out on the console? > > VERBOSE=1 gmake Thanks. That was helpfull. From brian at freeswitch.org Fri Jun 26 19:27:56 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 21:27:56 -0500 Subject: [Freeswitch-users] Version 1.0.4 not working with custom channel variables? In-Reply-To: References: Message-ID: <3372AF70-46CB-4B71-8455-B8E1719D22D0@freeswitch.org> Execute the app verbose_events /b On Jun 26, 2009, at 9:10 PM, Adam Wilt wrote: > Hi, I recently tried upgrading FreeSWITCH from 1.0.2 to 1.0.4pre8 on > my server, but the application I wrote uses custom channel variables > which I create using "setVariable" in Python. In version 1.0.2, I am > able to retrieve that variable within the event xml (it prepends the > variable name with variable_). But in version 1.0.4pre8, these > variables are missing from the xml events. Could this be a bug? > Thanks! > > From jason at jasonjgw.net Fri Jun 26 19:31:02 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 12:31:02 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090627003140.GA12881@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> <20090627003140.GA12881@jdc.jasonjgw.net> Message-ID: <20090627023102.GA21650@jdc.jasonjgw.net> Sorry - I misread the question. The error is: freeswitch at default> load mod_portaudio -ERR [module load file routine returned an error] 2009-06-27 12:30:00.740316 [CRIT] switch_loadable_module.c:871 Error Loading mod ule /opt/freeswitch/mod/mod_portaudio.so **/opt/freeswitch/mod/mod_portaudio.so: undefined symbol: snd_config** freeswitch at default> There is no error during the building. From lon at kickasspixels.com Fri Jun 26 19:50:21 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 26 Jun 2009 19:50:21 -0700 Subject: [Freeswitch-users] Event socket chunks don't always end with 2 line breaks? Message-ID: <5d3e0dc60906261950l60af9635r964049ed3ac976d8@mail.gmail.com> In the latest SVN this seem to be true for CHANNEL_HANGUP, which appends the following after the normal line breaks. "Disconnected, goodbye.\n See you at ClueCon! http://www.cluecon.com/" Perhaps it can be changed to conform the chunked data structure? variable_disconnect_message: Disconnected, goodbye. See you at ClueCon! http://www.cluecon.com Just a thought. Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/01b9c7a5/attachment-0002.html From brian at freeswitch.org Fri Jun 26 19:54:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 21:54:19 -0500 Subject: [Freeswitch-users] Event socket chunks don't always end with 2 line breaks? In-Reply-To: <5d3e0dc60906261950l60af9635r964049ed3ac976d8@mail.gmail.com> References: <5d3e0dc60906261950l60af9635r964049ed3ac976d8@mail.gmail.com> Message-ID: <2D44EBE1-115A-4305-AF37-BDAF85084271@freeswitch.org> The protocol has a content-length header.. you read exactly that many bytes and you won't be confused by this disconnect message. /b On Jun 26, 2009, at 9:50 PM, Lon Baker wrote: > In the latest SVN this seem to be true for CHANNEL_HANGUP, which > appends the following after the normal line breaks. > > "Disconnected, goodbye.\n > See you at ClueCon! http://www.cluecon.com/" > > Perhaps it can be changed to conform the chunked data structure? > > variable_disconnect_message: Disconnected, goodbye. See you at > ClueCon! http://www.cluecon.com > > Just a thought. > > Lon > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/0929f5b3/attachment-0002.html From lon at kickasspixels.com Fri Jun 26 20:06:54 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 26 Jun 2009 20:06:54 -0700 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 270 In-Reply-To: References: Message-ID: <5d3e0dc60906262006v7aad7b5dx549d6739898f48a4@mail.gmail.com> Brian, Thanks! I was just about to post a "never mind" after further reading source. Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/53b22cd0/attachment-0002.html From mike at jerris.com Fri Jun 26 20:19:02 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 26 Jun 2009 23:19:02 -0400 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090627021744.GA89233@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> <20090627021744.GA89233@quark.hightek.org> Message-ID: <0CFF5BFD-E500-4E8A-9802-F6BA0519B522@jerris.com> On Jun 26, 2009, at 10:17 PM, Vincent Stemen wrote: > On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: >> >> Can you post a bug to Jira.freeswitch.org with all these warnings, >> even better with patches to fix it. > > OK. I think I have narrowed the problem down to 3 issues. > > 1. The build system is treating even a single warning as a critical > error > and aborting compilation in that directory. As intended. > 2. Compilation continues into the next directory even though > compilation of > the previous directory was aborted, as you can see in the make > output below > where it went on to build "features". This can cause a chain > reaction of > other errors because stuff it expects to be there did not get > built in > previous stages. I suspect that is also the source of the > 'symbolic link' > error I was getting. I did not get that on this last compilation > after > fixing some of the warnings (see below). It can also make it > possible to > get to the end of the build and not know that stuff did not get > compiled > further back, leaving the package incomplete. > I've looked for this one and have not been able to nail it down. I must be missing an || exit somewhere in the module makefiles? Everytime I go to reproduce this issue I can't reproduce it. > 3. Lots of "return makes pointer from integer without a cast" > warnings > throughout the sofia-sip tree. This one, of course, is not the > actual show > stopper but is triggering the above problems and needs cleaned up > none the > less. I am sure these are trivial enough to fix but am a bit puzzled why I don't see them on. Any other platform. What version of gcc is this? Does dragonfly patch gcc to report more warnings than other platforms? > > > ================= > > Making all in su > LTCOMPILE su.lo > LTCOMPILE su_errno.lo > LTCOMPILE su_addrinfo.lo > LTCOMPILE su_alloc.lo > su_alloc.c: In function `sub_alloc': > su_alloc.c:428: warning: return makes pointer from integer without a > cast > su_alloc.c:511: warning: return makes pointer from integer without a > cast > su_alloc.c: In function `su_home_new': > su_alloc.c:555: warning: return makes pointer from integer without a > cast > su_alloc.c:557: warning: return makes pointer from integer without a > cast > su_alloc.c: In function `su_home_clone': > su_alloc.c:730: warning: return makes pointer from integer without a > cast > su_alloc.c:732: warning: return makes pointer from integer without a > cast > su_alloc.c: In function `su_realloc': > su_alloc.c:1315: warning: return makes pointer from integer without > a cast > su_alloc.c:1319: warning: return makes pointer from integer without > a cast > su_alloc.c: In function `su_salloc': > su_alloc.c:1518: warning: return makes pointer from integer without > a cast > gmake[9]: *** [su_alloc.lo] Error 1 > gmake[8]: *** [all] Error 2 > Making all in features > ... > > ================= > I confirmed the abort on warning issue (1) by fixing all warnings in > su_alloc.c. As you can see below, it went past it just fine until > it got > a warning in su_sprintf.c. > ================= > > Making all in su > LTCOMPILE su.lo > LTCOMPILE su_errno.lo > LTCOMPILE su_addrinfo.lo > LTCOMPILE su_alloc.lo > LTCOMPILE su_alloc_lock.lo > LTCOMPILE su_strdup.lo > LTCOMPILE su_sprintf.lo > su_sprintf.c: In function `su_vsprintf': > su_sprintf.c:98: warning: return makes pointer from integer without > a cast > gmake[9]: *** [su_sprintf.lo] Error 1 > gmake[8]: *** [all] Error 2 > Making all in features > ... > > ================= > The issue below of saying it was successfull, when it was not, is > apparently > part of issue 2. It continued into the "build" directory after the > previous > errors. > ================= > > Making all in packages > gmake[6]: *** [all-recursive] Error 1 > gmake[5]: *** [all] Error 2 > gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ > freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- > ua.la] Error 2 > gmake[3]: *** [mod_sofia-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + gmake install + > +----------------------------------------------+ > gmake[1]: *** [all-recursive] Error 1 > gmake: *** [all] Error 2 > > ================= > > I went ahead and established an account on jira.freeswitch.org. I > see there > are separate projects for "FreeSWITCH-Buildsystem" and "sofia-sip". I > am guessing I should post two separate reports, one for issues 1 and > 2 for the > Buildsystem and one in sofia-sip for issue 3 since all the warnings > seem to be > from that code. > > I do not know the build system well enough to solve issues 1 and 2 > at this > time. However, I already have a patch I will provide for su_alloc.c > and can > continue working on resolving more of the casting warnings if > somebody else can > work on the build issue, which is the real show stopper. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Fri Jun 26 20:30:10 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 26 Jun 2009 20:30:10 -0700 Subject: [Freeswitch-users] att_xfer w/uuid Message-ID: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> I'm trying to use xml_rpc to initiate an att_xfer on channel A (which is already bridged to channel B), but I'm running into some issues. I know the uuid's from both channel A and B, but the documentation I found on att_xfer only seems to indicate a way to do this from DMTF presses occurring on channel A. my idea was to use xml_rpc to execute a lua script which would take a uuid as an argv and bind to the session with freeswitch.session(uuid). I tried this, but the audio breaks up with the session that the lua script binded too. Does anyone have any recommendations on how I might accomplish an assisted transfer w/o DTMF presses and bind_meta_app knowing only a uuid? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/2e47aa3e/attachment-0002.html From brian at freeswitch.org Fri Jun 26 21:08:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 26 Jun 2009 23:08:16 -0500 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> Message-ID: <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> Not sure what you want to do is doable via XML RPC. That app is to be run on an existing session. The other solution is to take both legs and park them.. Then execute bridge on one leg to the target transfer person. Once that call is up.. you can them park both of those.. uuid_bridge the two you wish to complete then hang up on the third one. I think if you just uuid_bridge the two you want in the end the third one will just hangup. /b On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: > I'm trying to use xml_rpc to initiate an att_xfer on channel A > (which is already bridged to channel B), but I'm running into some > issues. > > I know the uuid's from both channel A and B, but the documentation I > found on att_xfer only seems to indicate a way to do this from DMTF > presses occurring on channel A. > > my idea was to use xml_rpc to execute a lua script which would take > a uuid as an argv and bind to the session with > freeswitch.session(uuid). > > I tried this, but the audio breaks up with the session that the lua > script binded too. Does anyone have any recommendations on how I > might accomplish an assisted transfer w/o DTMF presses and > bind_meta_app knowing only a uuid? > > Thanks. > > --matt From mattdfong at gmail.com Fri Jun 26 21:30:43 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 26 Jun 2009 21:30:43 -0700 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> Message-ID: <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> can you 3 way with uuid_bridge? --matt On Fri, Jun 26, 2009 at 9:08 PM, Brian West wrote: > Not sure what you want to do is doable via XML RPC. That app is to be > run on an existing session. The other solution is to take both legs > and park them.. Then execute bridge on one leg to the target transfer > person. Once that call is up.. you can them park both of those.. > uuid_bridge the two you wish to complete then hang up on the third > one. I think if you just uuid_bridge the two you want in the end the > third one will just hangup. > > /b > > On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: > > > I'm trying to use xml_rpc to initiate an att_xfer on channel A > > (which is already bridged to channel B), but I'm running into some > > issues. > > > > I know the uuid's from both channel A and B, but the documentation I > > found on att_xfer only seems to indicate a way to do this from DMTF > > presses occurring on channel A. > > > > my idea was to use xml_rpc to execute a lua script which would take > > a uuid as an argv and bind to the session with > > freeswitch.session(uuid). > > > > I tried this, but the audio breaks up with the session that the lua > > script binded too. Does anyone have any recommendations on how I > > might accomplish an assisted transfer w/o DTMF presses and > > bind_meta_app knowing only a uuid? > > > > Thanks. > > > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090626/065b047d/attachment-0002.html From dome at tel.co.th Fri Jun 26 23:45:36 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 27 Jun 2009 13:45:36 +0700 Subject: [Freeswitch-users] How to cancel session in Javascript Message-ID: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Dear All, I try s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); if (s.ready()){ s.setVariable("nibble_rate", "2.5"); s.setVariable("nibble_account", "0838833133"); s.execute("nibblebill", "heartbeat 5"); bridge(session,s); }; my question is 1. How to cancel create s session (by dtmf ) like a * in bridge app 2. when i hangup before s session ready is posible to cancel ? Best Regards. Dome C. From jason at jasonjgw.net Sat Jun 27 04:00:35 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 27 Jun 2009 21:00:35 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090627023102.GA21650@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> <20090626103349.GA25435@jdc.jasonjgw.net> <4A44AB08.1050503@gmail.com> <20090627000541.GB7509@jdc.jasonjgw.net> <20090627003140.GA12881@jdc.jasonjgw.net> <20090627023102.GA21650@jdc.jasonjgw.net> Message-ID: <20090627110035.GA10140@jdc.jasonjgw.net> As a further note on this subject, temporarily downgrading to libtool 1.5.26 and rebuilding FreeSWITCH gave me a working mod_portaudio.so module. Obviously this doesn't solve the problem, but it does prove that, as suspected, the migration to libtool 2.2.6a was the cause. Any suggestions on how to track down the build system bug would be welcome. From msc at freeswitch.org Sat Jun 27 10:09:32 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 27 Jun 2009 10:09:32 -0700 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> Message-ID: <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> Couldn't you just throw all the calls into a conference at this point? -MC Sent from my iPhone On Jun 26, 2009, at 9:30 PM, Matthew Fong wrote: > can you 3 way with uuid_bridge? > > --matt > > On Fri, Jun 26, 2009 at 9:08 PM, Brian West > wrote: > Not sure what you want to do is doable via XML RPC. That app is to be > run on an existing session. The other solution is to take both legs > and park them.. Then execute bridge on one leg to the target transfer > person. Once that call is up.. you can them park both of those.. > uuid_bridge the two you wish to complete then hang up on the third > one. I think if you just uuid_bridge the two you want in the end the > third one will just hangup. > > /b > > On Jun 26, 2009, at 10:30 PM, Matthew Fong wrote: > > > I'm trying to use xml_rpc to initiate an att_xfer on channel A > > (which is already bridged to channel B), but I'm running into some > > issues. > > > > I know the uuid's from both channel A and B, but the documentation I > > found on att_xfer only seems to indicate a way to do this from DMTF > > presses occurring on channel A. > > > > my idea was to use xml_rpc to execute a lua script which would take > > a uuid as an argv and bind to the session with > > freeswitch.session(uuid). > > > > I tried this, but the audio breaks up with the session that the lua > > script binded too. Does anyone have any recommendations on how I > > might accomplish an assisted transfer w/o DTMF presses and > > bind_meta_app knowing only a uuid? > > > > Thanks. > > > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/95b6986e/attachment-0002.html From brian at freeswitch.org Sat Jun 27 16:05:58 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 27 Jun 2009 18:05:58 -0500 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> Message-ID: <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> my thinking exactly. /b On Jun 27, 2009, at 12:09 PM, Michael S Collins wrote: > Couldn't you just throw all the calls into a conference at this point? > -MC > > Sent from my iPhone From Mailings at kh-dev.de Sat Jun 27 16:47:08 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 28 Jun 2009 01:47:08 +0200 Subject: [Freeswitch-users] Result of an application... Message-ID: Hi, maybe a stupid question, but how can I find out the result of an application? If I do (in perl) $session->execute("bridge", "user/${DestNr}\@$VARS{domain}"); How do I know if this was successful or if the user was busy or if the phone doesn't exist? Is there any status variable for the result of an execute (or even any other command)? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/eba58ba1/attachment-0002.html From mattdfong at gmail.com Sat Jun 27 19:18:48 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Sat, 27 Jun 2009 19:18:48 -0700 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> Message-ID: <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> there's a reference on the wiki to a three_way dial plan command. What does that do? What's the best way to put 2 bridged callers into a new conference? Must I park both uuid's first, and then transfer both to an extension that will add them to a new conference? Is there a way to do this without any break in the audio? Thanks... --matt On Sat, Jun 27, 2009 at 4:05 PM, Brian West wrote: > my thinking exactly. > > /b > > On Jun 27, 2009, at 12:09 PM, Michael S Collins wrote: > > > Couldn't you just throw all the calls into a conference at this point? > > -MC > > > > Sent from my iPhone > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090627/1b1248c0/attachment-0002.html From jason at jasonjgw.net Sat Jun 27 19:32:37 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 28 Jun 2009 12:32:37 +1000 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> Message-ID: <20090628023237.GA7470@jdc.jasonjgw.net> Matthew Fong wrote: > What's the best way to put 2 bridged callers into a new conference? Must I > park both uuid's first, and then transfer both to an extension that will add > them to a new conference? No, it's uuid_transfer with the -both option to transfer both legs to the conference extension. From brian at freeswitch.org Sat Jun 27 19:39:27 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 27 Jun 2009 21:39:27 -0500 Subject: [Freeswitch-users] att_xfer w/uuid In-Reply-To: <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> References: <4256bf830906262030x3e6ddcb3h747d1c90d65f1d56@mail.gmail.com> <9C13F5D1-A804-4FF3-A7A0-FFB37BBDB074@freeswitch.org> <4256bf830906262130o385ee4b0w5893e63c9422a9a2@mail.gmail.com> <0056234A-AAF9-4A04-9A2F-F64DC980412F@freeswitch.org> <082875A7-A3FD-4414-8F48-8E96A2B2FAD5@freeswitch.org> <4256bf830906271918o5bade78dncfdc0de1071ae581@mail.gmail.com> Message-ID: <0EEF60D0-2FE1-424F-B1B5-5234C8EDC04F@freeswitch.org> yes and in your case you can forget it. With what you wanna do its not possible to use that. /b On Jun 27, 2009, at 9:18 PM, Matthew Fong wrote: > there's a reference on the wiki to a three_way dial plan command. > What does that do? > > What's the best way to put 2 bridged callers into a new conference? > Must I park both uuid's first, and then transfer both to an > extension that will add them to a new conference? Is there a way to > do this without any break in the audio? Thanks... > > --matt From vince.freeswitch at hightek.org Sat Jun 27 22:17:06 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Sun, 28 Jun 2009 00:17:06 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <0CFF5BFD-E500-4E8A-9802-F6BA0519B522@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> <20090627021744.GA89233@quark.hightek.org> <0CFF5BFD-E500-4E8A-9802-F6BA0519B522@jerris.com> Message-ID: <20090628051706.GA13252@quark.hightek.org> On Fri, Jun 26, 2009 at 11:19:02PM -0400, Michael Jerris wrote: > > > On Jun 26, 2009, at 10:17 PM, Vincent Stemen > wrote: > > > On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > >> > >> Can you post a bug to Jira.freeswitch.org with all these warnings, > >> even better with patches to fix it. > > > > OK. I think I have narrowed the problem down to 3 issues. > > > > 1. The build system is treating even a single warning as a critical > > error > > and aborting compilation in that directory. > > As intended. Interesting. I don't think I have seen any other projects force the build to fail on warnings. Although, I think that is better than allowing warnings to accumulate and never get cleaned up like I see on a lot of the projects out there. > > 2. Compilation continues into the next directory even though > > compilation of > > the previous directory was aborted, as you can see in the make > > output below > > where it went on to build "features". This can cause a chain > > reaction of > > other errors because stuff it expects to be there did not get > > built in > > previous stages. I suspect that is also the source of the > > 'symbolic link' > > error I was getting. I did not get that on this last compilation > > after > > fixing some of the warnings (see below). It can also make it > > possible to > > get to the end of the build and not know that stuff did not get > > compiled > > further back, leaving the package incomplete. > > > > I've looked for this one and have not been able to nail it down. I > must be missing an || exit somewhere in the module makefiles? > Everytime I go to reproduce this issue I can't reproduce it. Since you are aware of it and working on it, do you still want me to take the time to create a problem report on jira for this? If so, do you want to add a platform option for Dragonfly BSD first? Or should I just select FreeBSD? > > 3. Lots of "return makes pointer from integer without a cast" > > warnings > > throughout the sofia-sip tree. This one, of course, is not the > > actual show > > stopper but is triggering the above problems and needs cleaned up > > none the > > less. > > I am sure these are trivial enough to fix but am a bit puzzled why I > don't see them on. Any other platform. What version of gcc is this? > Does dragonfly patch gcc to report more warnings than other platforms? Hmm.. It could be that you guys are all running gcc-4 (?). I am working on a bit older installation that is running gcc 3.4.6. Although, I was told on the dragonfly irc that gcc-4 generates more warnings by default than gcc-3. So I don't know. They said they did not know of any special gcc configurations or patches for gcc on Dragonfly. There was one thing unanimous though. Everybody I spoke with on the #c and the #dragonflybsd irc did not like the way the lines are coded that are generating most of the errors :-). e.g. return (void)(errno = EINVAL), NULL; I went ahead and posted a bug report on jira under the sofia-sip project, with a patch that fixes all the warnings for the first file (su_alloc.c). More patches will follow. I went ahead and selected FreeBSD as the platform. I thought I would point that out in case you guys want to add Dragonfly BSD and change the platform on this issue report. From nicolas at medularis.com Sun Jun 28 09:16:49 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Sun, 28 Jun 2009 12:16:49 -0400 Subject: [Freeswitch-users] hangup_cause NONE vs. NORMAL_CLEARING Message-ID: <1b46b4e80906280916g5fb065e9w65183f9117609f68@mail.gmail.com> I have a small JS script that makes a call, plays a sound file and then hangs up. For each call it makes, I log the hangup_cause variable on the CHANNEL_HANGUP_COMPLETE event. Most of the time, when calls are successful, I get a NORMAL_CLEARING cause, but sometimes I'll get a NONE cause. I wanted to know what the difference between these two is, because there is no reference to NONE in the wiki (http://wiki.freeswitch.org/wiki/Hangup_causes ). Thanks, Nicol?s -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/9b5cc05b/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Jun 28 11:49:18 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 19:49:18 +0100 Subject: [Freeswitch-users] Confused with event content lengths Message-ID: Hi Guys, I'm trying to parse events in C++ for an outbound socket. The docs are a little contradictory, so I wonder if someone could help me out. As I understand it and event is terminated with double LF's (\n\n) However if there is a Content-Length header the wiki very confusingly says 'Content-Length is the length of the event beginning AFTER the very next LF only line ("\n") and inclusive the trailing LF/LF pair ("\n\n")' BUT the example says it's after the \n\n in the header!! Which is it? In addition, it also looks like the event body is also terminated by a \n\n. If this is the case, why do I care about content length value, can't I simply read until I get the termination sequence? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/eaba5298/attachment-0002.html From jmesquita at gmail.com Sun Jun 28 11:57:21 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 28 Jun 2009 15:57:21 -0300 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: References: Message-ID: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> If I am not mistaken, you are always safe reading the amount data expressed on Content-Length since it is calculated based on the total message length before it is sent out of FS. >From a protocol point of view, it would indeed be much better to rely on something such as Content-Length then \n\n termination string. As I get to know more and more the core developers, I doubt they would rely on the latter. Hope it helps... jmesquita On Sun, Jun 28, 2009 at 3:49 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?m trying to parse events in C++ for an outbound socket. The docs are a > little contradictory, so I wonder if someone could help me out. > > > > As I understand it and event is terminated with double LF?s (\n\n) However > if there is a Content-Length header the wiki very confusingly says > > > > ?Content-Length is the length of the event beginning *AFTER* the very next > LF only line ("\n") and *inclusive* the trailing LF/LF pair ("\n\n")? > > > > BUT the example says it?s after the \n\n in the header!! Which is it? > > > > In addition, it also looks like the event body is also terminated by a > \n\n. If this is the case, why do I care about content length value, can?t > I simply read until I get the termination sequence? > > > > Regards, > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/cb93a7dc/attachment-0002.html From brian at freeswitch.org Sun Jun 28 12:23:28 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 28 Jun 2009 14:23:28 -0500 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> References: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> Message-ID: <15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> Yes it says 264 bytes read exactly 264 bytes or die trying. /b On Jun 28, 2009, at 1:57 PM, Jo?o Mesquita wrote: > If I am not mistaken, you are always safe reading the amount data > expressed on Content-Length since it is calculated based on the > total message length before it is sent out of FS. > > From a protocol point of view, it would indeed be much better to > rely on something such as Content-Length then \n\n termination > string. As I get to know more and more the core developers, I doubt > they would rely on the latter. > > Hope it helps... > > jmesquita From danishmoosa at gmail.com Sun Jun 28 04:30:44 2009 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Sun, 28 Jun 2009 17:30:44 +0600 Subject: [Freeswitch-users] bypass media mode is stateful? Message-ID: Hi, Although its obvious but just asking for confirmation :) Is bypass media mode in FS is fairly stateful. If it in huge network so its cdr(billing) is relaiable ? -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/a15b03d7/attachment-0002.html From brian at freeswitch.org Sun Jun 28 12:30:07 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 28 Jun 2009 14:30:07 -0500 Subject: [Freeswitch-users] bypass media mode is stateful? In-Reply-To: References: Message-ID: <77E11941-BD88-4CA9-83A2-D7C77376A190@freeswitch.org> Yes CDR is still valid.. the media is just bypassed... please review the FAQ. /b On Jun 28, 2009, at 6:30 AM, Muhammad Danish Moosa wrote: > Hi, > > Although its obvious but just asking for confirmation :) > > Is bypass media mode in FS is fairly stateful. If it in huge network > so its cdr(billing) is relaiable ? > From nik.middleton at noblesolutions.co.uk Sun Jun 28 12:38:01 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 20:38:01 +0100 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: <15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> References: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com> <15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> Message-ID: But from where? After the double LF of the header as one part of the wiki says or after the line containing the content-length that another part of the wiki says? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 28 June 2009 20:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Confused with event content lengths Yes it says 264 bytes read exactly 264 bytes or die trying. /b On Jun 28, 2009, at 1:57 PM, Jo?o Mesquita wrote: > If I am not mistaken, you are always safe reading the amount data > expressed on Content-Length since it is calculated based on the > total message length before it is sent out of FS. > > From a protocol point of view, it would indeed be much better to > rely on something such as Content-Length then \n\n termination > string. As I get to know more and more the core developers, I doubt > they would rely on the latter. > > Hope it helps... > > jmesquita _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Jun 28 15:40:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 23:40:20 +0100 Subject: [Freeswitch-users] Confused with event content lengths In-Reply-To: References: <5a8712120906281157o2168a531pb676112a3b8e3b86@mail.gmail.com><15C27898-0157-440C-8735-4864D35CD576@freeswitch.org> Message-ID: OK, finally figured it out. Have updated the Wiki to remove ambiguity and posted some SUDO code Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 28 June 2009 20:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Confused with event content lengths But from where? After the double LF of the header as one part of the wiki says or after the line containing the content-length that another part of the wiki says? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 28 June 2009 20:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Confused with event content lengths Yes it says 264 bytes read exactly 264 bytes or die trying. /b On Jun 28, 2009, at 1:57 PM, Jo?o Mesquita wrote: > If I am not mistaken, you are always safe reading the amount data > expressed on Content-Length since it is calculated based on the > total message length before it is sent out of FS. > > From a protocol point of view, it would indeed be much better to > rely on something such as Content-Length then \n\n termination > string. As I get to know more and more the core developers, I doubt > they would rely on the latter. > > Hope it helps... > > jmesquita _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Jun 28 15:55:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 28 Jun 2009 23:55:14 +0100 Subject: [Freeswitch-users] Myevents in outbound socket Message-ID: Hi Guys, I've almost got my c++ outbound socket control prog running, however even though the filter works, it would be truly great to just subscribe to myevents as even with the filter in place I get lots of channel Execute and complete events which I don't really need. Problem is, is that mod_VMD isn't included in those events, even though it is channel specific. Is there any chance that this will be included? If not, can someone point me to where myevents is defined and I'll have a go at it myself. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/9ac0b4db/attachment-0002.html From brian at freeswitch.org Sun Jun 28 15:59:56 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 28 Jun 2009 17:59:56 -0500 Subject: [Freeswitch-users] Myevents in outbound socket In-Reply-To: References: Message-ID: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> Are you using ESL? /b On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote: > Hi Guys, > > I?ve almost got my c++ outbound socket control prog running, however > even though the filter works, it would be truly great to just > subscribe to myevents as even with the filter in place I get lots of > channel Execute and complete events which I don?t really need. > Problem is, is that mod_VMD isn?t included in those events, even > though it is channel specific. Is there any chance that this will > be included? If not, can someone point me to where myevents is > defined and I?ll have a go at it myself. > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/2f02ac81/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Jun 28 16:14:21 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 00:14:21 +0100 Subject: [Freeswitch-users] Myevents in outbound socket In-Reply-To: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> References: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> Message-ID: Nope. Can't find much on the Wiki on how to interface with ESL using C++. I want to control the outbound socket from a windows 2003 server only because that's what I'm familiar with. Is there some portable C++ or C code? ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 29 June 2009 00:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Myevents in outbound socket Are you using ESL? /b On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote: Hi Guys, I've almost got my c++ outbound socket control prog running, however even though the filter works, it would be truly great to just subscribe to myevents as even with the filter in place I get lots of channel Execute and complete events which I don't really need. Problem is, is that mod_VMD isn't included in those events, even though it is channel specific. Is there any chance that this will be included? If not, can someone point me to where myevents is defined and I'll have a go at it myself. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/5ff0a126/attachment-0002.html From jmesquita at gmail.com Sun Jun 28 16:46:22 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 28 Jun 2009 20:46:22 -0300 Subject: [Freeswitch-users] Myevents in outbound socket In-Reply-To: References: <2E6A4A59-A76A-4F41-9132-3E98F5BB61A7@freeswitch.org> Message-ID: <5a8712120906281646ma945857h942f72fff4c25ac2@mail.gmail.com> You should definitely look at ESL, dude. Take a look at ${SVNROOT}/libs/esl/. There is a esl_oop inside that might give you a go. Beware that this is only an interface for SWIG, but might be useful to you if you extend it. Later, jmesquita On Sun, Jun 28, 2009 at 8:14 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Nope. > > > > Can?t find much on the Wiki on how to interface with ESL using C++. I want > to control the outbound socket from a windows 2003 server only because > that?s what I?m familiar with. Is there some portable C++ or C code? > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* 29 June 2009 00:00 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Myevents in outbound socket > > > > Are you using ESL? > > > > /b > > > > On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote: > > > > Hi Guys, > > > > I?ve almost got my c++ outbound socket control prog running, however even > though the filter works, it would be truly great to just subscribe to > myevents as even with the filter in place I get lots of channel Execute and > complete events which I don?t really need. Problem is, is that mod_VMD > isn?t included in those events, even though it is channel specific. Is > there any chance that this will be included? If not, can someone point me > to where myevents is defined and I?ll have a go at it myself. > > > > Regards, > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090628/d2cdbbfc/attachment-0002.html From yudha2008 at gmail.com Sun Jun 28 21:53:15 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 29 Jun 2009 10:23:15 +0530 Subject: [Freeswitch-users] javascript FIFO (First In First Out) Message-ID: *Hi, I have configured inbound through JavaScript it is working well. Through dialplan i have configured FIFO it is also working fine but i want to configure FIFO (First In First Out) through JavaScript. Is there any link or examples to configure FIFO through JavaScript which will assist me to resolve this problem. can any one assist me to do this above process. Thanks in advance. Thanks with Regards, N.Baskar. * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/49b52cdb/attachment-0002.html From mprabhuram at gmail.com Mon Jun 29 00:23:58 2009 From: mprabhuram at gmail.com (Prabhuram Mohan) Date: Mon, 29 Jun 2009 00:23:58 -0700 Subject: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC Message-ID: Hi All, Abode AIR based on flash player can be used to connect to freeswitch and issue commands through XMLRPC. I read the internet to do this plumbing and also got help through generous fellow developers from AIR & freeswitch community. Now that it is successfully done, here is the complete code for the benefit of people who are following suite. Comments are welcome!! Code available here - http://neoalchemist.tumblr.com/post/132134683 Thanks Prabhu From shaheryarkh at googlemail.com Mon Jun 29 00:42:27 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 29 Jun 2009 08:42:27 +0100 Subject: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC In-Reply-To: References: Message-ID: Good job man! This is really useful. Thank you. On Mon, Jun 29, 2009 at 8:23 AM, Prabhuram Mohan wrote: > Hi All, > > Abode AIR based on flash player can be used to connect to freeswitch > and issue commands through XMLRPC. I read the internet to do this > plumbing and also got help through generous fellow developers from AIR > & freeswitch community. Now that it is successfully done, here is the > complete code for the benefit of people who are following suite. > Comments are welcome!! > > Code available here - http://neoalchemist.tumblr.com/post/132134683 > > Thanks > Prabhu > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/80018d38/attachment-0002.html From jingwei.yang at gmail.com Mon Jun 29 02:25:09 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 29 Jun 2009 17:25:09 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906241923k1d5cc1b6qc8025781fc7cfd09@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> Message-ID: <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> Hi Chris, any thoughts? Thanks, -Jingwei On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang wrote: > Hi Chris, here's the one that confuses me. As far as I understand, the > extension 888 defined in public.xml is for picking up incoming calls. It > should have no influence on outgoing calls, right? If not, what is to write > to fit my case? (originate dingaling/gmail.com/userAAA at gmail.com&bridge(dingaling/ > gmail.com/userBBB at gmail.com), both userAAA and userBBB can be internal or > external). > > Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm > not quite sure what to include. So I make it very simple. > > > > > > > > Here are three relative parameters in client.xml: > > > > > > Still, I got no echo for internal Ip calls. Please let me know where goes > wrong. > > Thanks, > -Jingwei > > On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen wrote: > >> I guess you have the problem here, >> in client.xml you have >> >> >> but you only define extension 888 in default context, >> that's why nobody can reach you from public. >> >> under /usr/local/freeswitch/conf/dialplan >> >> define extension 888 in public.xml to the proper extension you expect, and >> check the console log from fs_cli when you do gtalk calling to your gmail >> client, you will find out the solution to your issue. >> >> chris >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/da9fddaf/attachment-0002.html From darklion11 at yahoo.com Mon Jun 29 03:44:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 03:44:41 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? Message-ID: <24251951.post@talk.nabble.com> Hi, Is there any available license G729 for freeswitch? I need it to match G729 of Asterisks? If any please help me and instructions how to install this thing. Thanks -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24251951.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From chris.chen2004 at gmail.com Mon Jun 29 03:46:13 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Mon, 29 Jun 2009 06:46:13 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> Message-ID: <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> Jingwei, I don't know if you have the 888 defined in default.xml? also you have to define $${domain}. please do " dl_debug on" from fs_cli, and watch the console logs and see what's going on when you try calling from external. Most likely your dialplan is not correctly defined. Chris On Mon, Jun 29, 2009 at 5:25 AM, Jingwei Yang wrote: > Hi Chris, any thoughts? > > Thanks, > -Jingwei > > > On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang wrote: > >> Hi Chris, here's the one that confuses me. As far as I understand, the >> extension 888 defined in public.xml is for picking up incoming calls. It >> should have no influence on outgoing calls, right? If not, what is to write >> to fit my case? (originate dingaling/gmail.com/userAAA at gmail.com&bridge(dingaling/ >> gmail.com/userBBB at gmail.com), both userAAA and userBBB can be internal or >> external). >> >> Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm >> not quite sure what to include. So I make it very simple. >> >> >> >> >> >> >> >> Here are three relative parameters in client.xml: >> >> >> >> >> >> Still, I got no echo for internal Ip calls. Please let me know where goes >> wrong. >> >> Thanks, >> -Jingwei >> >> On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen wrote: >> >>> I guess you have the problem here, >>> in client.xml you have >>> >>> >>> but you only define extension 888 in default context, >>> that's why nobody can reach you from public. >>> >>> under /usr/local/freeswitch/conf/dialplan >>> >>> define extension 888 in public.xml to the proper extension you expect, >>> and check the console log from fs_cli when you do gtalk calling to your >>> gmail client, you will find out the solution to your issue. >>> >>> chris >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/eb19bc01/attachment-0002.html From jason at jasonjgw.net Mon Jun 29 03:51:13 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 29 Jun 2009 20:51:13 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24251951.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> Message-ID: <20090629105113.GA4756@jdc.jasonjgw.net> Edmar Cruz wrote: > > Is there any available license G729 for freeswitch? Yes. It was announced here a few days ago - see the list archives. From darklion11 at yahoo.com Mon Jun 29 04:01:02 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 04:01:02 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <20090629105113.GA4756@jdc.jasonjgw.net> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> Message-ID: <24252099.post@talk.nabble.com> Yup. I got this mid_dahdi_codec but instructions how to install it? Jason White-14 wrote: > > Edmar Cruz wrote: >> >> Is there any available license G729 for freeswitch? > > Yes. It was announced here a few days ago - see the list archives. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24252099.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From odermann at googlemail.com Mon Jun 29 05:30:11 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 29 Jun 2009 14:30:11 +0200 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> Message-ID: <5e414ed0906290530q2fdb0643k34d9f883dc52deec@mail.gmail.com> thanks for your answers. i did not know, that muxing is cpu intensive. i just thought, that it would not matter, if one is muxing 50/50 or 30/70. for playing back a soundfile, while one can hear the other end it seems, that muxing is required. so the level of muxing should not make a difference!? anyway, if there is no other/better way, we have to do it with sox. no, we are not using stereo-files. kind regards dennis From javieraristizabal at gmail.com Mon Jun 29 06:47:05 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Mon, 29 Jun 2009 08:47:05 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24252099.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> Message-ID: mid_dahdi_codec is the source to build the TC400B card. On Mon, Jun 29, 2009 at 6:01 AM, Edmar Cruz wrote: > > Yup. I got this mid_dahdi_codec but instructions how to install it? > > Jason White-14 wrote: >> >> Edmar Cruz wrote: >>> >>> ? Is there any available license G729 for freeswitch? >> >> Yes. It was announced here a few days ago - see the list archives. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24252099.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Jun 29 06:51:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 08:51:42 -0500 Subject: [Freeswitch-users] mux with uuid_displace: soundfile to loud In-Reply-To: <5e414ed0906290530q2fdb0643k34d9f883dc52deec@mail.gmail.com> References: <5e414ed0906260438j14da8546t7b5103194b26c9f8@mail.gmail.com> <6D9C9BC7-0072-492D-B22B-F2F337BA46AC@jerris.com> <87f2f3b90906261046s1f4240adw45082db1a872711b@mail.gmail.com> <8B149722-D853-4517-BC5E-3ACE2318DF46@freeswitch.org> <5e414ed0906290530q2fdb0643k34d9f883dc52deec@mail.gmail.com> Message-ID: <39237CF4-51E2-4044-8302-24DF77DEBF50@freeswitch.org> Yes muxing like this will cause the volume to go up a little bit depending on the source input file. /b On Jun 29, 2009, at 7:30 AM, Dennis wrote: > thanks for your answers. i did not know, that muxing is cpu intensive. > i just thought, that it would not matter, if one is muxing 50/50 or > 30/70. for playing back a soundfile, while one can hear the other end > it seems, that muxing is required. so the level of muxing should not > make a difference!? > > anyway, if there is no other/better way, we have to do it with sox. > > no, we are not using stereo-files. > > > kind regards > dennis From mike at jerris.com Mon Jun 29 08:06:50 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:06:50 -0400 Subject: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC In-Reply-To: References: Message-ID: <481FFA3C-C0F8-4B32-A4DE-D53498D2B547@jerris.com> If you are interested, we can create a contrib directory for you in the codebase and you can commit it right in tree. If interested, please log on to irc and we can coordinate setting up an account for you. Mike On Jun 29, 2009, at 3:23 AM, Prabhuram Mohan wrote: > Hi All, > > Abode AIR based on flash player can be used to connect to freeswitch > and issue commands through XMLRPC. I read the internet to do this > plumbing and also got help through generous fellow developers from AIR > & freeswitch community. Now that it is successfully done, here is the > complete code for the benefit of people who are following suite. > Comments are welcome!! > > Code available here - http://neoalchemist.tumblr.com/post/132134683 > > Thanks > Prabhu > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shiyanov at gmail.com Mon Jun 29 08:07:57 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 29 Jun 2009 19:07:57 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: I've tried with the snapshot (06.26.2009) - and situation had become even worse - now both agents hear nothing.. Maybe problem is in my sip_profiles? Here they are: http://pastebin.freeswitch.org/pastebin.php?dl=9510 http://pastebin.freeswitch.org/pastebin.php?dl=9511 On Fri, Jun 26, 2009 at 10:03 PM, Brian West wrote: > Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk > also... due to the lines below. > > /b > > On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: > > > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 > > 192.168.147.130 > > s=FreeSWITCH > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/19a5dd14/attachment-0002.html From sz.krisz at freemail.hu Sun Jun 28 18:36:46 2009 From: sz.krisz at freemail.hu (szentesik) Date: Sun, 28 Jun 2009 18:36:46 -0700 (PDT) Subject: [Freeswitch-users] CTI In-Reply-To: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> Message-ID: <24247328.post@talk.nabble.com> Maxim Tsvetov wrote: > > Hello! > > We are seeking possibilities to use CTI features with Freeswitch. > > This features are: > - click-to-dial > - call popup > - answer call,hangup > - call transfer > > > Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, > CSTA..) > or there is already written module or third-party software? > ... > Currently working on some CSTA support (http://cstainside.sourceforge.net/). The MakeCall, DeliveredEvent, ClearConnection, TransferCall/SingleStepTransfer things required for the features above are on the list, the AnswerCall implementation is open (I'm not sure whether the FreeSWITCH is able to answer calls for any of the SIP clients available). -- View this message in context: http://www.nabble.com/CTI-tp24094686p24247328.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From fdenkens at ilibris.be Mon Jun 29 08:11:55 2009 From: fdenkens at ilibris.be (Frederik Denkens) Date: Mon, 29 Jun 2009 17:11:55 +0200 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation Message-ID: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> Hi, Following the recommendations of this list, we went with an external gateway to connect to the BRI based ISDN network. We'd like to configure a Patoon 4554 2xBRI <-> SIP gateway with Freeswitch to connect to the public ISDN network. The freeswitch config seems quite straightforward, but we don't manage to get the Patton to register with the Freeswitch and vise versa. Does anybody have some insight/sample config/tips they can share with us on this? Honesty obliges me to say that our experience with the Patton product is quite limited, which is not a big help for such a complex product. Many thanks in advance! Frederik. From msc at freeswitch.org Mon Jun 29 08:17:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 08:17:14 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24252099.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> Message-ID: <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> On Mon, Jun 29, 2009 at 4:01 AM, Edmar Cruz wrote: > > Yup. I got this mid_dahdi_codec but instructions how to install it? > Buy TC400B Install it and the TC400B drivers Load mod_dahdi_codec Enjoy low quality G729 calls. :) -MC > > Jason White-14 wrote: > > > > Edmar Cruz wrote: > >> > >> Is there any available license G729 for freeswitch? > > > > Yes. It was announced here a few days ago - see the list archives. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Is-there-any-license-G729--tp24251951p24252099.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/65d16b54/attachment-0002.html From brian at freeswitch.org Mon Jun 29 08:19:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 10:19:16 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> Message-ID: Everyone has this need for lower bandwidth calls... I tend to march the other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a single ulaw call) /b On Jun 29, 2009, at 10:17 AM, Michael Collins wrote: > > > On Mon, Jun 29, 2009 at 4:01 AM, Edmar Cruz > wrote: > > Yup. I got this mid_dahdi_codec but instructions how to install it? > > Buy TC400B > Install it and the TC400B drivers > Load mod_dahdi_codec > Enjoy low quality G729 calls. :) > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/76e563d3/attachment-0002.html From brian at freeswitch.org Mon Jun 29 08:20:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 10:20:22 -0500 Subject: [Freeswitch-users] CTI In-Reply-To: <24247328.post@talk.nabble.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> Message-ID: Nice are you the project leader? /b On Jun 28, 2009, at 8:36 PM, szentesik wrote: > Currently working on some CSTA support (http://cstainside.sourceforge.net/ > ). > The MakeCall, DeliveredEvent, ClearConnection, > TransferCall/SingleStepTransfer things required for the features > above are > on the list, the AnswerCall implementation is open (I'm not sure > whether the > FreeSWITCH is able to answer calls for any of the SIP clients > available). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/051eb4fd/attachment-0002.html From mike at jerris.com Mon Jun 29 08:38:25 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:38:25 -0400 Subject: [Freeswitch-users] Result of an application... In-Reply-To: References: Message-ID: <548806F2-CE53-4D6F-89F8-C0EC7304C941@jerris.com> The application interface doesn't return a status like that. Different applications may set channel variables on an app by app basis. if your doing a bridge you can look at for example: http://wiki.freeswitch.org/wiki/Channel_Variables#originate_disposition Mike On Jun 27, 2009, at 7:47 PM, Klaus Hochlehnert wrote: > Hi, > > maybe a stupid question, but how can I find out the result of an > application? > > If I do (in perl) $session->execute("bridge", "user/${DestNr}\@ > $VARS{domain}"); > > How do I know if this was successful or if the user was busy or if > the phone doesn?t exist? > Is there any status variable for the result of an execute (or even > any other command)? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/93996e91/attachment-0002.html From mike at jerris.com Mon Jun 29 08:40:24 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:40:24 -0400 Subject: [Freeswitch-users] hangup_cause NONE vs. NORMAL_CLEARING In-Reply-To: <1b46b4e80906280916g5fb065e9w65183f9117609f68@mail.gmail.com> References: <1b46b4e80906280916g5fb065e9w65183f9117609f68@mail.gmail.com> Message-ID: <68622E7A-8512-4631-8875-3745F85DC956@jerris.com> I would be curious if you have debug output of a call that is returning NONE there. That seems like we should be setting a hang-up cause somewhere. Mike On Jun 28, 2009, at 12:16 PM, Nicolas Brenner wrote: > I have a small JS script that makes a call, plays a sound file and > then hangs up. For each call it makes, I log the hangup_cause > variable on the CHANNEL_HANGUP_COMPLETE event. Most of the time, > when calls are successful, I get a NORMAL_CLEARING cause, but > sometimes I'll get a NONE cause. I wanted to know what the > difference between these two is, because there is no reference to > NONE in the wiki (http://wiki.freeswitch.org/wiki/Hangup_causes). > > Thanks, > > Nicol?s > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/2dd8a8be/attachment-0002.html From shiyanov at gmail.com Mon Jun 29 08:41:28 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Mon, 29 Jun 2009 19:41:28 +0400 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: Update again: FS debug logs of the problematic part http://pastebin.freeswitch.org/pastebin.php?dl=9512 Artem On Mon, Jun 29, 2009 at 7:07 PM, Artem Shiyanov wrote: > I've tried with the snapshot (06.26.2009) - and situation had become even > worse - now both agents hear nothing.. > Maybe problem is in my sip_profiles? > Here they are: > http://pastebin.freeswitch.org/pastebin.php?dl=9510 > http://pastebin.freeswitch.org/pastebin.php?dl=9511 > > > > > On Fri, Jun 26, 2009 at 10:03 PM, Brian West wrote: > >> Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk >> also... due to the lines below. >> >> /b >> >> On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote: >> >> > o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 >> > 192.168.147.130 >> > s=FreeSWITCH >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/0804e9af/attachment-0002.html From mike at jerris.com Mon Jun 29 08:42:16 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:42:16 -0400 Subject: [Freeswitch-users] javascript FIFO (First In First Out) In-Reply-To: References: Message-ID: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> Fifo is not configured through javascript, it is configured via the configuration files. http://wiki.freeswitch.org/wiki/Mod_fifo Mike On Jun 29, 2009, at 12:53 AM, Baskar wrote: > Hi, > > I have configured inbound through JavaScript it is working well. > > Through dialplan i have configured FIFO it is also working fine but > i want to configure FIFO (First In First Out) through JavaScript. > > Is there any link or examples to configure FIFO through JavaScript > which will assist me to resolve this problem. > > can any one assist me to do this above process. > > Thanks in advance. > > Thanks with Regards, > N.Baskar. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/b4d33be5/attachment-0002.html From william.suffill at gmail.com Mon Jun 29 08:42:52 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 29 Jun 2009 11:42:52 -0400 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> Message-ID: <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> You always have a way of beating to a different drum but whatever works. Good to see that the options are available and the user can choose what's best for their unique situation. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/3088057d/attachment-0002.html From mike at jerris.com Mon Jun 29 08:43:54 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 11:43:54 -0400 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation In-Reply-To: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> References: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> Message-ID: Posting a quick sip trace of the failed register will probably be helpful. Also, are the debug logs any help? Mike On Jun 29, 2009, at 11:11 AM, Frederik Denkens wrote: > Hi, > > Following the recommendations of this list, we went with an external > gateway to connect to the BRI based ISDN network. > > We'd like to configure a Patoon 4554 2xBRI <-> SIP gateway with > Freeswitch to connect to the public ISDN network. The freeswitch > config seems quite straightforward, but we don't manage to get the > Patton to register with the Freeswitch and vise versa. > > Does anybody have some insight/sample config/tips they can share with > us on this? Honesty obliges me to say that our experience with the > Patton product is quite limited, which is not a big help for such a > complex product. From brian at freeswitch.org Mon Jun 29 08:44:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 10:44:52 -0500 Subject: [Freeswitch-users] one-way audio after playback+bridge In-Reply-To: References: Message-ID: <72397F6D-609C-43C4-998C-B5968506BF66@freeswitch.org> Now you'll need to outline step by step what you're doing to reproduce this problem. /b On Jun 29, 2009, at 10:41 AM, Artem Shiyanov wrote: > Update again: > FS debug logs of the problematic part > http://pastebin.freeswitch.org/pastebin.php?dl=9512 > > Artem > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/00784a0d/attachment-0002.html From mnhassan at usa.net Mon Jun 29 08:48:34 2009 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 29 Jun 2009 22:48:34 +0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> Message-ID: <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> Hi, I just noted Micheal Collins mention "Enjoy lower quality G.729 calls". By "lower quality" do you mean G.729 in TC400B is lower in quality compared to software compression? Or is that comparing G.729 with G.711? Regards HASSAN On Mon, Jun 29, 2009 at 10:42 PM, William Suffill wrote: > You always have a way of beating to a different drum but whatever works. > > Good to see that the options are available and the user can choose what's > best for their unique situation. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From christian.loeschenkohl at xpirio.com Mon Jun 29 08:52:27 2009 From: christian.loeschenkohl at xpirio.com (=?UTF-8?B?Q2hyaXN0aWFuIEzDtnNjaGVua29obA==?=) Date: Mon, 29 Jun 2009 17:52:27 +0200 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation In-Reply-To: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> References: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> Message-ID: <4A48E33B.8040002@xpirio.com> hi i'm inalp patton certified, so maybe i can help if you can post your config (export startup config). please also describe your setup if i could get http+telnet access to the smartnode, the system would be registered in an minute br On 2009-06-29 17:11, Frederik Denkens wrote: > Hi, > > Following the recommendations of this list, we went with an external > gateway to connect to the BRI based ISDN network. > > We'd like to configure a Patoon 4554 2xBRI<-> SIP gateway with > Freeswitch to connect to the public ISDN network. The freeswitch > config seems quite straightforward, but we don't manage to get the > Patton to register with the Freeswitch and vise versa. > > Does anybody have some insight/sample config/tips they can share with > us on this? Honesty obliges me to say that our experience with the > Patton product is quite limited, which is not a big help for such a > complex product. > > Many thanks in advance! > > Frederik. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From mike at jerris.com Mon Jun 29 09:00:25 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 12:00:25 -0400 Subject: [Freeswitch-users] Patton 4554 as gateway for a FreeSwitch installation In-Reply-To: <4A48E33B.8040002@xpirio.com> References: <71233AC3-D140-48DA-B18D-D752E92D1B36@ilibris.be> <4A48E33B.8040002@xpirio.com> Message-ID: If you come up with a good configuration, we would appretiate it if someone could post this to the wiki. Mike On Jun 29, 2009, at 11:52 AM, Christian L?schenkohl wrote: > hi > > i'm inalp patton certified, so maybe i can help > if you can post your config (export startup config). > please also describe your setup > if i could get http+telnet access to the smartnode, the system would > be registered in an minute > > br > From steveu at coppice.org Mon Jun 29 09:09:59 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 00:09:59 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> Message-ID: <4A48E757.60103@coppice.org> Nyamul Hassan wrote: > Hi, > > I just noted Micheal Collins mention "Enjoy lower quality G.729 > calls". By "lower quality" do you mean G.729 in TC400B is lower in > quality compared to software compression? Or is that comparing G.729 > with G.711? > I think he is just saddened by the way people tolerant crappy quality, and how slow the takeup of wideband voice has been. The TC400B doesn't do G.729. It does G.729A, which is significantly lower in quality. However, G.729A is what almost all equipment that vaguely says G.729 actually implements. Its lousy, but few people care. The TC400B works as well or as badly as anything else. Steve From jmesquita at gmail.com Mon Jun 29 09:22:04 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 29 Jun 2009 13:22:04 -0300 Subject: [Freeswitch-users] CTI In-Reply-To: References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> Message-ID: <5a8712120906290922q78515102t32e2f23dc4093015@mail.gmail.com> I am interested to know more about this. Are you using ESL to then translate CSTA calls to FS? Wouldn't this be great to be added as an FS module as an alternative to ESL? It would enable lots of existing CTI applications to work with FS. jmesquita On Mon, Jun 29, 2009 at 12:20 PM, Brian West wrote: > Nice are you the project leader? > /b > > On Jun 28, 2009, at 8:36 PM, szentesik wrote: > > Currently working on some CSTA support (http://cstainside.sourceforge.net/ > ). > The MakeCall, DeliveredEvent, ClearConnection, > TransferCall/SingleStepTransfer things required for the features above are > on the list, the AnswerCall implementation is open (I'm not sure whether > the > FreeSWITCH is able to answer calls for any of the SIP clients available). > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/03b422fe/attachment-0002.html From mnhassan at usa.net Mon Jun 29 09:42:23 2009 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 29 Jun 2009 23:42:23 +0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48E757.60103@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> Message-ID: <9618647f0906290942s277ea8b2q6012b583d13d5f3e@mail.gmail.com> For wholesale termination setups, I think it makes it much more cost-effective to implement G.729 or even G.723. They can never be the same quality as G.711, but it saves bandwidth by multiple whole numbers, and is therefore, more sought after in markets where BW is expensive. One a side note, we were thinking of deploying a FS solution for our new retail platform. We are looking for a vendor who can provide a very good billing platform. Can you recommend me some? We are willing to pay for the service, but would rather go with an opensource project that provides paid support. Thank you in advance for your suggestions. Regards HASSAN On Mon, Jun 29, 2009 at 11:09 PM, Steve Underwood wrote: > Nyamul Hassan wrote: >> Hi, >> >> I just noted Micheal Collins mention "Enjoy lower quality G.729 >> calls". ?By "lower quality" do you mean G.729 in TC400B is lower in >> quality compared to software compression? ?Or is that comparing G.729 >> with G.711? >> > I think he is just saddened by the way people tolerant crappy quality, > and how slow the takeup of wideband voice has been. > > The TC400B doesn't do G.729. It does G.729A, which is significantly > lower in quality. However, G.729A is what almost all equipment that > vaguely says G.729 actually implements. Its lousy, but few people care. > The TC400B works as well or as badly as anything else. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From tayeb.meftah at gmail.com Mon Jun 29 09:46:53 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 29 Jun 2009 16:46:53 +0000 Subject: [Freeswitch-users] skypiax featur request Message-ID: <4A48EFFD.3080905@gmail.com> hello, this message is for skypiax developers / contributor please i have to request: i know a software named "SiSky" this is a SIP to Skype software gateway that let you interconnect your SIP IpPbx to the Skype network if i dial a skype speed dial number from my IpPhone, skype will no to bi visible but skypiax: if i dial any number i get skype visible and i heare the ringing sound in skype cool you try to hide it in the next release? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4196 (20090629) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From brian at freeswitch.org Mon Jun 29 09:49:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 11:49:27 -0500 Subject: [Freeswitch-users] skypiax featur request In-Reply-To: <4A48EFFD.3080905@gmail.com> References: <4A48EFFD.3080905@gmail.com> Message-ID: Please contact the author or open a jira as a feature request. /b On Jun 29, 2009, at 11:46 AM, Meftah Tayeb wrote: > hello, > this message is for skypiax developers / contributor > please i have to request: > i know a software named "SiSky" > this is a SIP to Skype software gateway that let you interconnect your > SIP IpPbx to the Skype network > if i dial a skype speed dial number from my IpPhone, skype will no > to bi > visible > but skypiax: if i dial any number i get skype visible and i heare the > ringing sound in skype > cool you try to hide it in the next release? > thanks From msc at freeswitch.org Mon Jun 29 09:52:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 09:52:33 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48E757.60103@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> Message-ID: <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> On Mon, Jun 29, 2009 at 9:09 AM, Steve Underwood wrote: > Nyamul Hassan wrote: > > Hi, > > > > I just noted Micheal Collins mention "Enjoy lower quality G.729 > > calls". By "lower quality" do you mean G.729 in TC400B is lower in > > quality compared to software compression? Or is that comparing G.729 > > with G.711? > > > I think he is just saddened by the way people tolerant crappy quality, > and how slow the takeup of wideband voice has been. > > The TC400B doesn't do G.729. It does G.729A, which is significantly > lower in quality. However, G.729A is what almost all equipment that > vaguely says G.729 actually implements. Its lousy, but few people care. > The TC400B works as well or as badly as anything else. > > Steve > Well said! Like Steve has pointed out in the past: G729/G729A is a race to the bottom. After using WB and UWB codecs all day every day for the past 6 months I just can't live with G729 or even GSM for that matter. However, to each his own. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/b1fc84e2/attachment-0002.html From brian at freeswitch.org Mon Jun 29 10:00:09 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 12:00:09 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> Cellphones have really lowered everyones expectations of what quality is. I think if each of us in the US could stab our Cellphone providers in the neck and get away with it.. I know I would... YES AT&T I'm talking about your sorry excuse for service.... /b On Jun 29, 2009, at 11:52 AM, Michael Collins wrote: > Well said! Like Steve has pointed out in the past: G729/G729A is a > race to the bottom. After using WB and UWB codecs all day every day > for the past 6 months I just can't live with G729 or even GSM for > that matter. However, to each his own. > -MC > From steveu at coppice.org Mon Jun 29 10:11:51 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 01:11:51 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> Message-ID: <4A48F5D7.3050405@coppice.org> Brian West wrote: > Cellphones have really lowered everyones expectations of what quality > is. I think if each of us in the US could stab our Cellphone > providers in the neck and get away with it.. I know I would... YES > AT&T I'm talking about your sorry excuse for service.... > The 3G cellular standards have wideband features (that's what AMR-WB was developed for), but few (maybe no) operators enable them. I assume all UMTS phones actually support wideband, but unless the network is prepared to accept negotiation for it, phone support isn't much use. The networks would rather push proven flops like video calls. Steve From brian at freeswitch.org Mon Jun 29 10:19:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 12:19:16 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48F5D7.3050405@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> Message-ID: <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> Video calls would be great if any network would actually SUPPORT IT! (speaking US only) /b On Jun 29, 2009, at 12:11 PM, Steve Underwood wrote: > The 3G cellular standards have wideband features (that's what AMR-WB > was > developed for), but few (maybe no) operators enable them. I assume all > UMTS phones actually support wideband, but unless the network is > prepared to accept negotiation for it, phone support isn't much use. > The > networks would rather push proven flops like video calls. > > Steve From steveu at coppice.org Mon Jun 29 10:31:29 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 01:31:29 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> Message-ID: <4A48FA71.3050406@coppice.org> Video calls are a really really bad idea. People who think otherwise really haven't thought about it at all. They are available here, and people desperately don't want them to be. Brian West wrote: > Video calls would be great if any network would actually SUPPORT IT! > (speaking US only) > > /b > > On Jun 29, 2009, at 12:11 PM, Steve Underwood wrote: > > >> The 3G cellular standards have wideband features (that's what AMR-WB >> was >> developed for), but few (maybe no) operators enable them. I assume all >> UMTS phones actually support wideband, but unless the network is >> prepared to accept negotiation for it, phone support isn't much use. >> The >> networks would rather push proven flops like video calls. >> >> Steve Steve From brian at freeswitch.org Mon Jun 29 10:36:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 12:36:08 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48FA71.3050406@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> Message-ID: <50E06D00-0525-4BD2-8404-3654390EB0A9@freeswitch.org> Cuz its harder to lie to your wife! :P /b On Jun 29, 2009, at 12:31 PM, Steve Underwood wrote: > Video calls are a really really bad idea. People who think otherwise > really haven't thought about it at all. They are available here, and > people desperately don't want them to be. From msc at freeswitch.org Mon Jun 29 10:40:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 10:40:51 -0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Message-ID: <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> What kind of application are you building? Usually you want to use the dialplan to initiate the call and then let the js do the logical heavy lifting. -MC On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost wrote: > Dear All, > > I try > > s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); > if (s.ready()){ > s.setVariable("nibble_rate", "2.5"); > s.setVariable("nibble_account", "0838833133"); > s.execute("nibblebill", "heartbeat 5"); > bridge(session,s); > }; > > my question is > 1. How to cancel create s session (by dtmf ) like a * in bridge app > 2. when i hangup before s session ready is posible to cancel ? > > Best Regards. > > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a0e9273b/attachment-0002.html From steveu at coppice.org Mon Jun 29 10:45:31 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 30 Jun 2009 01:45:31 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <50E06D00-0525-4BD2-8404-3654390EB0A9@freeswitch.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <50E06D00-0525-4BD2-8404-3654390EB0A9@freeswitch.org> Message-ID: <4A48FDBB.80900@coppice.org> Brian West wrote: > Cuz its harder to lie to your wife! :P > > /b > > On Jun 29, 2009, at 12:31 PM, Steve Underwood wrote: > > >> Video calls are a really really bad idea. People who think otherwise >> really haven't thought about it at all. They are available here, and >> people desperately don't want them to be. >> Now you're thinking :-) Steve From Chr.Schaefers at gmx.de Mon Jun 29 10:51:43 2009 From: Chr.Schaefers at gmx.de (chschaef) Date: Mon, 29 Jun 2009 10:51:43 -0700 (PDT) Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? Message-ID: <24228390.post@talk.nabble.com> >Hi, >there is a "edit online" option to use to change the option on the phone. >You have to first install SIP Voip Settings Tool from the Nokia forum, and >then change the option "Secure call" to prefered from NCT. >Ognjen Hello, did you actually try it and succeed? The description in the Nokia forum is existing but the implementation of the listener is apparently missing ... regards, chschaef -- View this message in context: http://www.nabble.com/Anybody-tried-Nokia-E71-%28symbian-S60-3rd%29-with-Pjsip-and-TLS-SRTP--tp23230269p24228390.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 29 11:08:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 13:08:40 -0500 Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? In-Reply-To: <24228390.post@talk.nabble.com> References: <24228390.post@talk.nabble.com> Message-ID: You MUST use TLS btw for SRTP to work. I'm working right now to interop with my E63 /b On Jun 29, 2009, at 12:51 PM, chschaef wrote: > Hello, > > did you actually try it and succeed? The description in the Nokia > forum is > existing but the implementation of the listener is apparently > missing ... > > regards, > chschaef > -- From dave at 3c.co.uk Mon Jun 29 11:16:48 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 29 Jun 2009 21:16:48 +0300 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A48FA71.3050406@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> Message-ID: <1246299408.3877.24.camel@dk-d820> On Tue, 2009-06-30 at 01:31 +0800, Steve Underwood wrote: > Video calls are a really really bad idea. People who think otherwise > really haven't thought about it at all. They are available here, and > people desperately don't want them to be. Video calls between 3G phones have been available in the UK for some years. I've only ever made two using one of these to real people, both across a table in a pub to show them what it looked like; I have made goodness knows how many to an IVR while trying to pick 3G-324M apart. However, there are some instances where they're very useful. For example, my family is geographically quite dispersed, and we use Skype with video a lot - particularly for grandparents to keep up with grandchildren. The usefulness and appropriateness of video calling depends very much on the target market; it's not, of itself, a bad thing. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From sz.krisz at freemail.hu Mon Jun 29 11:25:27 2009 From: sz.krisz at freemail.hu (szentesik) Date: Mon, 29 Jun 2009 11:25:27 -0700 (PDT) Subject: [Freeswitch-users] CTI In-Reply-To: <5a8712120906290922q78515102t32e2f23dc4093015@mail.gmail.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> <5a8712120906290922q78515102t32e2f23dc4093015@mail.gmail.com> Message-ID: <24259342.post@talk.nabble.com> The FS side CSTA "server" (mod_csta_socket) works as a loadable module exactly as you described, practically adding native CSTA support to it. (I test it only on Windows yet, but the code was made to let it build on any FS platform if someone put some effort into writing the makefiles etc.) Krisztian Jo?o Mesquita-3 wrote: > > I am interested to know more about this. Are you using ESL to then > translate > CSTA calls to FS? Wouldn't this be great to be added as an FS module as an > alternative to ESL? It would enable lots of existing CTI applications to > work with FS. > > jmesquita > > On Mon, Jun 29, 2009 at 12:20 PM, Brian West wrote: > >> Nice are you the project leader? >> /b >> >> On Jun 28, 2009, at 8:36 PM, szentesik wrote: >> >> Currently working on some CSTA support >> (http://cstainside.sourceforge.net/ >> ). > ... > -- View this message in context: http://www.nabble.com/CTI-tp24094686p24259342.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Jun 29 11:27:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 11:27:05 -0700 Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? In-Reply-To: References: <24228390.post@talk.nabble.com> Message-ID: <87f2f3b90906291127x60e630edu7c7d6b7992aca0c4@mail.gmail.com> On Mon, Jun 29, 2009 at 11:08 AM, Brian West wrote: > You MUST use TLS btw for SRTP to work. I'm working right now to > interop with my E63 > Sweet! Let us know when it's done. That's very cool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/164d5e40/attachment-0002.html From Chr.Schaefers at gmx.de Mon Jun 29 11:36:16 2009 From: Chr.Schaefers at gmx.de (chschaef) Date: Mon, 29 Jun 2009 11:36:16 -0700 (PDT) Subject: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? In-Reply-To: References: <49F2D874.7070002@gmx.net> <20090425095738.GA27179@jdc.jasonjgw.net> <49F341DF.20206@gmx.net> <4468a6770904270313l75bfd3fhc7406a526f001395@mail.gmail.com> <49F7138D.2060803@gmx.net> <4468a6770904280752s4f301ff6i5c966603b7b2d3f1@mail.gmail.com> <24228390.post@talk.nabble.com> Message-ID: <24259345.post@talk.nabble.com> Brian West-3 wrote: > >>You MUST use TLS btw for SRTP to work. I'm working right now to >> interop with my E63 >>/b > > I have configured TLS & SRTP on the Nokia E71 (and E51). That works fine to make encrypted calls. :-) When making a call to the Nokia, it does not respond on port 5061. On port 5060, it responds but only with UDP packets. Looking into the Nokia developer forum, I have found info from 2007, that the TLS listener is not yet implemented - but I could not get any confirmation by Nokia if that is still valid. So I appreciate that you get back to me if you can make the E63 calling securely in both directions. regards, chschaef -- View this message in context: http://www.nabble.com/Anybody-tried-Nokia-E71-%28symbian-S60-3rd%29-with-Pjsip-and-TLS-SRTP--tp23230269p24259345.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Mon Jun 29 12:47:23 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 20:47:23 +0100 Subject: [Freeswitch-users] bridge call from outbound socket Message-ID: Hi Guys, Is it possible to bridge to another destination while controlling a call via the outbound socket? In other words, I'm controlling a call using an outbound socket and at some point want to originate a new call leg and bridge the two. If it can't be done that way, I'm thinking I could originate the call using an inbound socket, grab the uuid and then call " api uuid_bridge " ? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/79a1b239/attachment-0002.html From msc at freeswitch.org Mon Jun 29 13:21:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 13:21:50 -0700 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: Message-ID: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> would you mind doing a pb of your script that is handling the OB event socket connection? -MC On Mon, Jun 29, 2009 at 12:47 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Is it possible to bridge to another destination while controlling a call > via the outbound socket? > > > > In other words, I?m controlling a call using an outbound socket and at some > point want to originate a new call leg and bridge the two. > > > > If it can?t be done that way, I?m thinking I could originate the call using an inbound socket, grab the uuid and then call ? api uuid_bridge ? ? > > > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/695273c7/attachment-0002.html From sz.krisz at freemail.hu Mon Jun 29 13:34:42 2009 From: sz.krisz at freemail.hu (szentesik) Date: Mon, 29 Jun 2009 13:34:42 -0700 (PDT) Subject: [Freeswitch-users] CTI In-Reply-To: References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> Message-ID: <24261403.post@talk.nabble.com> Yes. It will also use to bring FS with a CTI application I'm a lead developer of closer, decided to make the integration open source/open standard based. Brian West-3 wrote: > > Nice are you the project leader? > > /b > > On Jun 28, 2009, at 8:36 PM, szentesik wrote: > >> Currently working on some CSTA support >> (http://cstainside.sourceforge.net/ >> ). >> The MakeCall, DeliveredEvent, ClearConnection, >> TransferCall/SingleStepTransfer things required for the features >> above are >> on the list, the AnswerCall implementation is open (I'm not sure >> whether the >> FreeSWITCH is able to answer calls for any of the SIP clients >> available). > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/CTI-tp24094686p24261403.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Jun 29 13:38:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 15:38:20 -0500 Subject: [Freeswitch-users] CTI In-Reply-To: <24261403.post@talk.nabble.com> References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> <24261403.post@talk.nabble.com> Message-ID: Are you interested in hosting any of it in our tree? /b On Jun 29, 2009, at 3:34 PM, szentesik wrote: > > Yes. It will also use to bring FS with a CTI application I'm a lead > developer > of closer, decided to make the integration open source/open standard > based. > From nik.middleton at noblesolutions.co.uk Mon Jun 29 13:43:54 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 21:43:54 +0100 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> Message-ID: Thanks, but I got it sorted. The bridge application along with the event-lock sorted. Works a treat, yippee! I'll write it up on the wiki Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 29 June 2009 21:22 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] bridge call from outbound socket would you mind doing a pb of your script that is handling the OB event socket connection? -MC On Mon, Jun 29, 2009 at 12:47 PM, Nik Middleton wrote: Hi Guys, Is it possible to bridge to another destination while controlling a call via the outbound socket? In other words, I'm controlling a call using an outbound socket and at some point want to originate a new call leg and bridge the two. If it can't be done that way, I'm thinking I could originate the call using an inbound socket, grab the uuid and then call " api uuid_bridge " ? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a17a2328/attachment-0002.html From msc at freeswitch.org Mon Jun 29 13:49:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 13:49:23 -0700 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> Message-ID: <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Thanks, but I got it sorted. The bridge application along with the event-lock > sorted. Works a treat, yippee! I?ll write it up on the wiki > > > Cool deal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a6b733db/attachment-0002.html From jmesquita at gmail.com Mon Jun 29 14:08:54 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 29 Jun 2009 18:08:54 -0300 Subject: [Freeswitch-users] CTI In-Reply-To: References: <89c9bbf80906180749j47c9e38bie22a7ef870f201e2@mail.gmail.com> <24247328.post@talk.nabble.com> <24261403.post@talk.nabble.com> Message-ID: <5a8712120906291408h54494e9fyde469901758bd757@mail.gmail.com> I would strongly suggest that. At least for the mod itself. That way, we can all contribute with it and keep it always compatible with the lib. jmesquita On Mon, Jun 29, 2009 at 5:38 PM, Brian West wrote: > Are you interested in hosting any of it in our tree? > > /b > > On Jun 29, 2009, at 3:34 PM, szentesik wrote: > > > > > Yes. It will also use to bring FS with a CTI application I'm a lead > > developer > > of closer, decided to make the integration open source/open standard > > based. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/eabe91ab/attachment-0002.html From marketing at cluecon.com Mon Jun 29 14:33:43 2009 From: marketing at cluecon.com (Michael Collins) Date: Mon, 29 Jun 2009 14:33:43 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Last Call For Early Birds! Message-ID: <87f2f3b90906291433m4819f096nf38e20e7c0cc673f@mail.gmail.com> Good news! It's not too late to sign up for the early bird special of $499 for ClueCon, but you'll need to act fast - July 1 is the last day for the early bird special and it's only a few days away. If you haven't already registered for ClueCon then please call us and do so right away. Expedia.com still has deals on hotel rooms but the Wyndham is filling up quickly so don't delay in getting your room booked. In other news we have a new media sponsor: Biz-News.com. They will be conducting interviews with some of the speakers this year and are helping to promote the event. Please drop by their site and check them out! Looking forward to seeing everybody this August! -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/e3eb9e23/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Jun 29 14:56:04 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 29 Jun 2009 22:56:04 +0100 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> Message-ID: One little annoyance though, I cannot for the life of me get a ringback tone while the B leg is ringing, I've tried putting ringback=${us-ring} in the originate params, but no deal, just silence until the call is answered. Anyone care to shed some light on this? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 29 June 2009 21:49 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] bridge call from outbound socket On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton wrote: Thanks, but I got it sorted. The bridge application along with the event-lock sorted. Works a treat, yippee! I'll write it up on the wiki Cool deal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/ceb84138/attachment-0002.html From msc at freeswitch.org Mon Jun 29 15:06:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 15:06:48 -0700 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> Message-ID: <87f2f3b90906291506n216db20du2de86d56593b96fe@mail.gmail.com> can you pb your script? -MC On Mon, Jun 29, 2009 at 2:56 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > One little annoyance though, I cannot for the life of me get a ringback > tone while the B leg is ringing, > > > > I?ve tried putting ringback=${us-ring} in the originate params, but no > deal, just silence until the call is answered. Anyone care to shed some > light on this? > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 29 June 2009 21:49 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] bridge call from outbound socket > > > > > > On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Thanks, but I got it sorted. The bridge application along with the event-lock > sorted. Works a treat, yippee! I?ll write it up on the wiki > > > > Cool deal. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/5be7eba8/attachment-0002.html From mike at jerris.com Mon Jun 29 15:10:16 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 29 Jun 2009 18:10:16 -0400 Subject: [Freeswitch-users] bridge call from outbound socket In-Reply-To: References: <87f2f3b90906291321n7f43b118ga8af45410ddb9589@mail.gmail.com> <87f2f3b90906291349r771e86dcn6b5a39671b9da7f@mail.gmail.com> Message-ID: I don't think we expand vars here, you will need to expand us-ring yourself and pass the string Mike On Jun 29, 2009, at 5:56 PM, Nik Middleton wrote: > One little annoyance though, I cannot for the life of me get a > ringback tone while the B leg is ringing, > > I?ve tried putting ringback=${us-ring} in the originate params, but > no deal, just silence until the call is answered. Anyone care to > shed some light on this? > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: 29 June 2009 21:49 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] bridge call from outbound socket > > > On Mon, Jun 29, 2009 at 1:43 PM, Nik Middleton > wrote: > Thanks, but I got it sorted. The bridge application along with the > event-lock sorted. Works a treat, yippee! I?ll write it up on the > wiki > > > > Cool deal. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/9111f146/attachment-0002.html From sz.krisz at freemail.hu Mon Jun 29 15:17:49 2009 From: sz.krisz at freemail.hu (=?ISO-8859-2?Q?Szentesi_Kriszti=E1n?=) Date: Tue, 30 Jun 2009 00:17:49 +0200 (CEST) Subject: [Freeswitch-users] CTI In-Reply-To: Message-ID: Yes. The project itself consist of 3 parts, the CSTAInsideCore static library for CSTA standard classes, the mod_csta_socket contains the FS specific stuff and a client program for learning purposes. Both the mod_csta_socket and the client links with the core library, all is standard C++. It is trivial to host the mod_csta_socket with FreeSWITCH, and would also agree to move the core library itself. Help from some experienced developers are always welcome. Krisztian Brian West ?rta: > Are you interested in hosting any of it in our tree? > > /b > > On Jun 29, 2009, at 3:34 PM, szentesik wrote: > > > > > Yes. It will also use to bring FS with a CTI application I'm a lead > > developer > > of closer, decided to make the integration open source/open standard > > based. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ________________________________________________________
Angol, n?met, spanyol, olasz, francia ny?ri intenz?v nyelvtanfolyamok indulnak j?nius 22-t?l a Bonus Nyelviskol?ban
http://ad.adverticum.net/b/cl,1,6022,335167,414321/click.prm -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/24ae026c/attachment-0002.html From msc at freeswitch.org Mon Jun 29 16:13:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 16:13:30 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 Message-ID: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> The FreeSWITCH development team is please to announce that there's a new version of FreeSWITCH available. Please update as soon as you reasonable can. More details available here: http://www.freeswitch.org/node/195 We appreciate everyone's help in making FreeSWITCH better. Please keep testing and reporting back! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/c761a15c/attachment-0002.html From jason at jasonjgw.net Mon Jun 29 17:18:58 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 10:18:58 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> Message-ID: <20090630001858.GA12969@jdc.jasonjgw.net> Brian West wrote: > Everyone has this need for lower bandwidth calls... I tend to march the > other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a > single ulaw call) 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. This also works well in 48khz conferences. I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, G.722, G.722.1 and (my current favourite) Celt all the way. From dujinfang at gmail.com Mon Jun 29 18:13:20 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 09:13:20 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: > Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/ Linux/Windows, and I'd like to know some UAs supporting WB and UWB codecs. Thanks. > Well said! Like Steve has pointed out in the past: G729/G729A is a > race to the bottom. After using WB and UWB codecs all day every day > for the past 6 months I just can't live with G729 or even GSM for > that matter. However, to each his own. > -MC From sprice at gmail.com Mon Jun 29 18:18:16 2009 From: sprice at gmail.com (SP) Date: Mon, 29 Jun 2009 20:18:16 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> FreeSWITCH runs on a Mac and can be configured as a UA... also supports WB and UWB! On Mon, Jun 29, 2009 at 20:13, seven wrote: > > > > Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/ > Linux/Windows, and I'd like to know some UAs supporting WB and UWB > codecs. > > Thanks. > > > Well said! Like Steve has pointed out in the past: G729/G729A is a > > race to the bottom. After using WB and UWB codecs all day every day > > for the past 6 months I just can't live with G729 or even GSM for > > that matter. However, to each his own. > > -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/09c1d335/attachment-0002.html From brian at freeswitch.org Mon Jun 29 18:21:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:21:25 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> Message-ID: oh and SUPER DUPER WIDE BAND. :P /b On Jun 29, 2009, at 8:18 PM, SP wrote: > FreeSWITCH runs on a Mac and can be configured as a UA... also > supports WB and UWB! From jason at jasonjgw.net Mon Jun 29 18:28:07 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 11:28:07 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> References: <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> Message-ID: <20090630012807.GA21806@jdc.jasonjgw.net> SP wrote: > FreeSWITCH runs on a Mac and can be configured as a UA... also supports WB > and UWB! Correct. It's especially good for those of us who prefer to avoid WIMP user interfaces. From craig at overthewire.com.au Mon Jun 29 18:26:15 2009 From: craig at overthewire.com.au (Craig Askings) Date: Tue, 30 Jun 2009 11:26:15 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Message-ID: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White : > Brian West wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! ?(btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From brian at freeswitch.org Mon Jun 29 18:31:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:31:39 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <20090630012807.GA21806@jdc.jasonjgw.net> References: <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> <20090630012807.GA21806@jdc.jasonjgw.net> Message-ID: <15355D4E-D2DE-4CAF-8DD1-8A587C7C6428@freeswitch.org> Its perfect for you isn't it... I rather love it too! ;) Simple... easy to use... and no fancy GUI to mess up... :P /b On Jun 29, 2009, at 8:28 PM, Jason White wrote: > Correct. It's especially good for those of us who prefer to avoid > WIMP user > interfaces. From dujinfang at gmail.com Mon Jun 29 18:36:43 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 09:36:43 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <7e2ac3270906291818t7b838a01pef22386c3104a3e0@mail.gmail.com> Message-ID: <68966E85-CACE-4F69-8909-FE5744FB542C@gmail.com> Sure I know, Just haven't tried that. :) Any way it only makes sense when both legs support WB or UWB, unfortunately most of our b-legs are PSTN :(. > FreeSWITCH runs on a Mac and can be configured as a UA... also > supports WB and UWB! From darklion11 at yahoo.com Mon Jun 29 18:37:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 18:37:59 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <1246299408.3877.24.camel@dk-d820> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <1246299408.3877.24.camel@dk-d820> Message-ID: <24261922.post@talk.nabble.com> Please give me some free drivers of G729... Please David Knell wrote: > > On Tue, 2009-06-30 at 01:31 +0800, Steve Underwood wrote: >> Video calls are a really really bad idea. People who think otherwise >> really haven't thought about it at all. They are available here, and >> people desperately don't want them to be. > > Video calls between 3G phones have been available in the UK for some > years. I've only ever made two using one of these to real people, both > across a table in a pub to show them what it looked like; I have made > goodness knows how many to an IVR while trying to pick 3G-324M apart. > > However, there are some instances where they're very useful. For > example, my family is geographically quite dispersed, and we use Skype > with video a lot - particularly for grandparents to keep up with > grandchildren. The usefulness and appropriateness of video calling > depends very much on the target market; it's not, of itself, a bad > thing. > > --Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24261922.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Mon Jun 29 18:41:15 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 11:41:15 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> Message-ID: <20090630014115.GA23733@jdc.jasonjgw.net> Craig Askings wrote: > Are there any hardware phones that support 48 Khz Celt and > automated/mass deployment? Actually... FreeSWITCH in a phone could be a very good project. The main obstacles are: 1. Someone would need to design and build the hardware, or find existing hardware that would be suitable. 2. A script would have to be written to control i/o so that the keyboard and display of the phone could be used to make calls. From brian at freeswitch.org Mon Jun 29 18:43:28 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:43:28 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24261922.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <1246299408.3877.24.camel@dk-d820> <24261922.post@talk.nabble.com> Message-ID: <294FD821-2768-4545-BC9A-9DA875FAB27E@freeswitch.org> Free is not possible. /b PS: and its a codec :P On Jun 29, 2009, at 8:37 PM, Edmar Cruz wrote: > > Please give me some free drivers of G729... Please From brian at freeswitch.org Mon Jun 29 18:45:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 20:45:02 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <20090630014115.GA23733@jdc.jasonjgw.net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> Message-ID: <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> Now who can we con into building an open platform thats nothing more than a linux box shaped like a nice looking phone? :P Video and Touch Screen are a requirement :) /b On Jun 29, 2009, at 8:41 PM, Jason White wrote: > > Actually... FreeSWITCH in a phone could be a very good project. > > The main obstacles are: > > 1. Someone would need to design and build the hardware, or find > existing > hardware that would be suitable. > > 2. A script would have to be written to control i/o so that the > keyboard and > display of the phone could be used to make calls. From craig at overthewire.com.au Mon Jun 29 18:58:45 2009 From: craig at overthewire.com.au (Craig Askings) Date: Tue, 30 Jun 2009 11:58:45 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> Message-ID: <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> Well Snom phones use linux. I wonder if the the Chinese factory that pumps them out does grey market versions. Unfortunately that doesn't meed the video, touch screen or even the nice looking requirements. Craig. p.s. Above thoughts are my own not my employers. 2009/6/30 Brian West : > Now who can we con into building an open platform thats nothing more > than a linux box shaped like a nice looking phone? ?:P > > Video and Touch Screen are a requirement :) > > /b > > On Jun 29, 2009, at 8:41 PM, Jason White wrote: > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From sprice at gmail.com Mon Jun 29 19:23:03 2009 From: sprice at gmail.com (SP) Date: Mon, 29 Jun 2009 21:23:03 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> Message-ID: <7e2ac3270906291923ka1578fewa1ea07857147d1e6@mail.gmail.com> I'm now imagining bkw with his iphone in front of the mirror. - touchscreen - check - video - check - nice looking - you know I am On Mon, Jun 29, 2009 at 20:58, Craig Askings wrote: > Well Snom phones use linux. I wonder if the the Chinese factory that > pumps them out does grey market versions. Unfortunately that doesn't > meed the video, touch screen or even the nice looking requirements. > > Craig. > > p.s. Above thoughts are my own not my employers. > > 2009/6/30 Brian West : > > Now who can we con into building an open platform thats nothing more > > than a linux box shaped like a nice looking phone? :P > > > > Video and Touch Screen are a requirement :) > > > > /b > > > > On Jun 29, 2009, at 8:41 PM, Jason White wrote: > > > > -- > Craig Askings > > Network Engineer | Over the Wire Pty Ltd > craig at overthewire.com.au | www.overthewire.com.au > Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/e1cdf8e5/attachment-0002.html From brian at freeswitch.org Mon Jun 29 19:30:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 21:30:11 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <7e2ac3270906291923ka1578fewa1ea07857147d1e6@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <7e2ac3270906291923ka1578fewa1ea07857147d1e6@mail.gmail.com> Message-ID: <0A6C17E1-6B63-44B4-8CBA-3339DD773030@freeswitch.org> iphone 3GS boi! /b On Jun 29, 2009, at 9:23 PM, SP wrote: > > I'm now imagining bkw with his iphone in front of the mirror. > touchscreen - check > video - check > nice looking - you know I am -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/a218a045/attachment-0002.html From drago at windstream.net Mon Jun 29 19:32:28 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 22:32:28 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> Message-ID: <019601c9f92b$036c02d0$0a440870$@net> Actually, Snom does have a version with color LCD touch screen - model 820. I'm not sure if it is in mass production yet. Regarding the "nice looking"... suit yourself :-) : http://www.snom.com/en/products/snom-820/ Drago -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Craig Askings Sent: Monday, June 29, 2009 9:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Well Snom phones use linux. I wonder if the the Chinese factory that pumps them out does grey market versions. Unfortunately that doesn't meed the video, touch screen or even the nice looking requirements. Craig. p.s. Above thoughts are my own not my employers. 2009/6/30 Brian West : > Now who can we con into building an open platform thats nothing more > than a linux box shaped like a nice looking phone? ?:P > > Video and Touch Screen are a requirement :) > > /b > > On Jun 29, 2009, at 8:41 PM, Jason White wrote: > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Jun 29 19:37:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 21:37:27 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <019601c9f92b$036c02d0$0a440870$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: BZZZT WRONG on the touch screen! /b On Jun 29, 2009, at 9:32 PM, Drago Totev wrote: > Actually, Snom does have a version with color LCD touch screen - > model 820. > I'm not sure if it is in mass production yet. > > Regarding the "nice looking"... suit yourself :-) : > http://www.snom.com/en/products/snom-820/ > > Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/d56708db/attachment-0002.html From zhaoxxqq at 163.com Mon Jun 29 19:42:21 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Tue, 30 Jun 2009 10:42:21 +0800 Subject: [Freeswitch-users] Question about outband event use java Message-ID: <200906301042207132716@163.com> Hi folks, I'm really newbee to touch outband event. I write socket below and I write java server code like below: import java.io.BufferedReader; import java.io.BufferedWriter; import java.io.IOException; import java.io.InputStreamReader; import java.io.OutputStreamWriter; import java.io.PrintWriter; import java.net.ServerSocket; import java.net.Socket; import java.util.regex.*; public class ServerCode { /** * @param args */ public static int portNo = 8023; public static void main(String[] args) throws IOException { // TODO code application logic here ServerSocket s= new ServerSocket(portNo); System.out.println("The Server is start:" + s); Socket socket=s.accept(); try{ System.out.println("Accept the Client:"+ socket); BufferedReader in =new BufferedReader(new InputStreamReader(socket.getInputStream())); PrintWriter out = new PrintWriter(new BufferedWriter(new OutputStreamWriter(socket.getOutputStream())),true); while(true){ String str = in.readLine(); if(str.equals("byebye")){ break; } System.out.println("In Server recieved the info:" + str); out.println("connect\n\n"); } } finally { System.out.println("close the server socket and the io."); socket.close(); s.close(); } } } When I use extension 1001 to dial 3001, the cli display below: 2009-06-30 18:35:27 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 at 61.55.140.138:5060 [b9c80e50-6561-11de-8972-f3830035270b] 2009-06-30 18:35:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1001->3001 in context default I have send "connect\n\n" to FS, but no response to outband server. Can anyone help me? 2009-06-30 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/d7cc94f5/attachment-0002.html From drago at windstream.net Mon Jun 29 19:48:32 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 22:48:32 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: <01af01c9f92d$419ef1f0$c4dcd5d0$@net> Nope. J I "touch" it on VoiceCon in Orlando this April. Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 29, 2009 10:37 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? BZZZT WRONG on the touch screen! /b On Jun 29, 2009, at 9:32 PM, Drago Totev wrote: Actually, Snom does have a version with color LCD touch screen - model 820. I'm not sure if it is in mass production yet. Regarding the "nice looking"... suit yourself :-) : http://www.snom.com/en/products/snom-820/ Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/6ff46e85/attachment-0002.html From jason at jasonjgw.net Mon Jun 29 19:50:28 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 12:50:28 +1000 Subject: [Freeswitch-users] Question about outband event use java In-Reply-To: <200906301042207132716@163.com> References: <200906301042207132716@163.com> Message-ID: <20090630025028.GA6448@jdc.jasonjgw.net> zhaoxxqq wrote: > and I write java server code like below: > > import java.io.BufferedReader; > import java.io.BufferedWriter; > import java.io.IOException; > import java.io.InputStreamReader; > import java.io.OutputStreamWriter; > import java.io.PrintWriter; > import java.net.ServerSocket; > import java.net.Socket; > import java.util.regex.*; Shouldn't you be using the Java event socket libraries that were discussed on the list recently? There's also one in the FreeSWITCH source tree now. From brian at freeswitch.org Mon Jun 29 19:51:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 21:51:18 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <01af01c9f92d$419ef1f0$c4dcd5d0$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <01af01c9f92d$419ef1f0$c4dcd5d0$@net> Message-ID: <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> Yah you can touch the screen all you want on the 820 but mine doesn't do jack! /b On Jun 29, 2009, at 9:48 PM, Drago Totev wrote: > Nope. J > > I ?touch? it on VoiceCon in Orlando this April. > > Drago > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/54b9fc35/attachment-0002.html From hads at nice.net.nz Mon Jun 29 19:55:32 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 30 Jun 2009 14:55:32 +1200 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: <1246330532.4148.3.camel@sodium> On Mon, 2009-06-29 at 21:37 -0500, Brian West wrote: > BZZZT WRONG on the touch screen! Just wrong on the model, it's the 870 which has a touch screen. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From drago at windstream.net Mon Jun 29 19:56:07 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 22:56:07 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <01af01c9f92d$419ef1f0$c4dcd5d0$@net> <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> Message-ID: <01c001c9f92e$5150ee40$f3f2cac0$@net> Correct. As I said - it is a prototype. We wasting time in this argument. Contact Snom if you are interested or. move on, buddy J Drago From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 29, 2009 10:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Yah you can touch the screen all you want on the 820 but mine doesn't do jack! /b On Jun 29, 2009, at 9:48 PM, Drago Totev wrote: Nope. J I "touch" it on VoiceCon in Orlando this April. Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/101d5857/attachment-0002.html From drago at windstream.net Mon Jun 29 20:01:05 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 23:01:05 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <1246330532.4148.3.camel@sodium> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> Message-ID: <01cb01c9f92f$029024f0$07b06ed0$@net> You are correct. My bad. Drago -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hadley Rich Sent: Monday, June 29, 2009 10:56 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? On Mon, 2009-06-29 at 21:37 -0500, Brian West wrote: > BZZZT WRONG on the touch screen! Just wrong on the model, it's the 870 which has a touch screen. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Jun 29 20:03:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 22:03:21 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <1246330532.4148.3.camel@sodium> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> Message-ID: <3FCFEAE4-94FC-45FD-813D-535896DE5ADD@freeswitch.org> I'll have to call Christian over at Snom and have him ship me one then! ;) /b On Jun 29, 2009, at 9:55 PM, Hadley Rich wrote: > Just wrong on the model, it's the 870 which has a touch screen. > > hads From brian at freeswitch.org Mon Jun 29 20:03:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 22:03:40 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <01c001c9f92e$5150ee40$f3f2cac0$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <01af01c9f92d$419ef1f0$c4dcd5d0$@net> <7719BE43-E3A8-4609-870C-BB6C8703AD05@freeswitch.org> <01c001c9f92e$5150ee40$f3f2cac0$@net> Message-ID: <7AD463C5-0A80-4625-81A1-038ED9EBB47E@freeswitch.org> I'll have to get one now... I have one of every other snom! :) /b On Jun 29, 2009, at 9:56 PM, Drago Totev wrote: > Correct. As I said ? it is a prototype. > > We wasting time in this argument. Contact Snom if you are interested > or? move on, buddy J > > Drago -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/f70077fe/attachment-0002.html From brian at freeswitch.org Mon Jun 29 20:04:12 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 29 Jun 2009 22:04:12 -0500 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <01cb01c9f92f$029024f0$07b06ed0$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> <01cb01c9f92f$029024f0$07b06ed0$@net> Message-ID: Its ok ... I have lysdexic moments almost daily. /b On Jun 29, 2009, at 10:01 PM, Drago Totev wrote: > You are correct. My bad. > > Drago From drago at windstream.net Mon Jun 29 20:07:44 2009 From: drago at windstream.net (Drago Totev) Date: Mon, 29 Jun 2009 23:07:44 -0400 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> <01cb01c9f92f$029024f0$07b06ed0$@net> Message-ID: <01d201c9f92f$f054d3c0$d0fe7b40$@net> Now I feel better! I thought it's me only... :-) Drago -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 29, 2009 11:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? Its ok ... I have lysdexic moments almost daily. /b On Jun 29, 2009, at 10:01 PM, Drago Totev wrote: > You are correct. My bad. > > Drago _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Mon Jun 29 20:08:40 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 11:08:40 +0800 Subject: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h Message-ID: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> Hi, I'm on the latest svn 14041, and I have tiff installed with "port install tiff", how can I tell FS to find libtiff at /opt/local/include? on Linux the package should be libtiff-dev of libtiff-devel, but I think tiff is the equivalent on Mac. previous versions build ok on my Mac, possible to include the libtiff in trunk like other libs? checking tiffio.h usability... no checking tiffio.h presence... no checking for tiffio.h... no checking pthread.h usability... yes checking pthread.h presence... yes checking for pthread.h... yes checking X11/X.h usability... yes checking X11/X.h presence... yes checking for X11/X.h... yes checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h usability... yes checking libxml/xmlmemory.h presence... yes checking for libxml/xmlmemory.h... yes checking libxml/parser.h usability... yes checking libxml/parser.h presence... yes checking for libxml/parser.h... yes checking libxml/xinclude.h usability... yes checking libxml/xinclude.h presence... yes checking for libxml/xinclude.h... yes checking FL/Fl.H usability... no checking FL/Fl.H presence... no checking for FL/Fl.H... no checking FL/Fl_Overlay_Window.H usability... no checking FL/Fl_Overlay_Window.H presence... no checking for FL/Fl_Overlay_Window.H... no checking FL/Fl_Light_Button.H usability... no checking FL/Fl_Light_Button.H presence... no checking for FL/Fl_Light_Button.H... no checking FL/fl_draw.H usability... no checking FL/fl_draw.H presence... no checking for FL/fl_draw.H... no checking FL/Fl_Cartesian.H usability... no checking FL/Fl_Cartesian.H presence... no checking for FL/Fl_Cartesian.H... no checking FL/Fl_Audio_Meter.H usability... no checking FL/Fl_Audio_Meter.H presence... no checking for FL/Fl_Audio_Meter.H... no checking for cos in -lm... yes checking for library containing sinf... none required checking for library containing cosf... none required checking for library containing tanf... none required checking for library containing asinf... none required checking for library containing acosf... none required checking for library containing atanf... none required checking for library containing atan2f... none required checking for library containing ceilf... none required checking for library containing floorf... none required checking for library containing powf... none required checking for library containing expf... none required checking for library containing logf... none required checking for library containing log10f... none required checking for TIFFOpen in -ltiff... no configure: error: "Can't build without libtiff (does your system require a libtiff-devel package?)" configure: error: ./configure.gnu failed for libs/spandsp From craig at overthewire.com.au Mon Jun 29 20:20:18 2009 From: craig at overthewire.com.au (Craig Askings) Date: Tue, 30 Jun 2009 13:20:18 +1000 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <1246330532.4148.3.camel@sodium> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> Message-ID: <8cc991dd0906292020u3528f483v1462e049877c664d@mail.gmail.com> Well the 8XX series is much prettier than the 3XX series. Though I wonder if anyone has thought of porting FreeSWITCH to the Palm Pre? Craig. 2009/6/30 Hadley Rich : > On Mon, 2009-06-29 at 21:37 -0500, Brian West wrote: >> BZZZT WRONG on the touch screen! > > Just wrong on the model, it's the 870 which has a touch screen. > > hads > -- > http://nicegear.co.nz > New Zealand's Open Source Hardware Supplier > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From diego.viola at gmail.com Mon Jun 29 20:26:36 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 29 Jun 2009 23:26:36 -0400 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <20090629105113.GA4756@jdc.jasonjgw.net> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> Message-ID: <86a32abc0906292026v27065787ida78f6b2ca0afb3c@mail.gmail.com> Look for this email: "[Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH" I think the development team is also about to offer G729 licensing if I'm not mistaken. On Mon, Jun 29, 2009 at 6:51 AM, Jason White wrote: > Edmar Cruz wrote: > > > > Is there any available license G729 for freeswitch? > > Yes. It was announced here a few days ago - see the list archives. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/34d1d2e5/attachment-0002.html From vince.freeswitch at hightek.org Mon Jun 29 21:19:12 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Mon, 29 Jun 2009 23:19:12 -0500 Subject: [Freeswitch-users] freeswitch on Dragonfly BSD (RLIMIT_AS issue) In-Reply-To: <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> References: <20090623040853.GA84157@quark.hightek.org> <191c3a030906230629kf1a9e07m79615485d1682768@mail.gmail.com> <20090624021530.GA749@quark.hightek.org> <4215D902-441A-4C01-AB49-12527F6E5C01@jerris.com> Message-ID: <20090630041912.GA2585@quark.hightek.org> On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: > > Can we make a patch ifdefing on RLIMIT_AS to make this always work > without patches to system header files? I have attached a patch to src/switch_core.c that fixes this. -------------- next part -------------- --- src/switch_core.c.orig 2009-06-29 19:29:22 -0700 +++ src/switch_core.c 2009-06-29 19:31:21 -0700 @@ -826,7 +826,9 @@ setrlimit(RLIMIT_RTPRIO, &rlp); #endif -#if !defined(__OpenBSD__) && !defined(__NetBSD__) +#if defined(__DragonFly__) + setrlimit(RLIMIT_VMEM, &rlp); +#elif !defined(__OpenBSD__) && !defined(__NetBSD__) setrlimit(RLIMIT_AS, &rlp); #endif #endif From vince.freeswitch at hightek.org Mon Jun 29 21:35:34 2009 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Mon, 29 Jun 2009 23:35:34 -0500 Subject: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD In-Reply-To: <20090623040853.GA84157@quark.hightek.org> References: <20090623040853.GA84157@quark.hightek.org> Message-ID: <20090630043534.GA2840@quark.hightek.org> Update: I tested compiling on my Dragonfly leaf development account which is running DragonFly 2.2.1-RELEASE with gcc-4.1.2. I did not get any of the cast warnings. It is apparently a gcc issue since I am running gcc-3.4.6 on my workstation with an older Dragonfly installation. Apparently gcc-3 does not like the "return x, y;" syntax that is used all over the sofia code. That is the source of most of the warnings (except 1 or 2 so far that just needed a cast). So, on DF 2.2.1, it appeared to compile almost successfully until near the end. It still has that sym link error and still has the build bug where it does not know there was an error and goes on to tell you it was successful. Here is the final output: making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /home/vince/freeswitch/freeswitch-20090623/libs/js/nsprpub/dist/include/nspr/.: Invalid argument gmake[9]: *** [export] Error 1 gmake[8]: *** [export] Error 2 gmake[7]: *** [export] Error 2 gmake[6]: *** [export] Error 2 gmake[5]: *** [/home/vince/freeswitch/freeswitch-20090623/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 From yj13535428332 at gmail.com Mon Jun 29 21:03:48 2009 From: yj13535428332 at gmail.com (jun yang) Date: Tue, 30 Jun 2009 12:03:48 +0800 Subject: [Freeswitch-users] how to record the conference manually? Message-ID: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> Hi,everyone. new to play with freeswitch. i have setup freeswitch and can call each other very well. now, a conference 3002 with several users in it. i want to record 3002 manually, but can't get the way. i have try fs_cli use the command: conference 3002 record /tmp/foo.wav it response: conference 3002 not found any clue? thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/faf33374/attachment-0002.html From brian at freeswitch.org Mon Jun 29 22:43:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 30 Jun 2009 00:43:30 -0500 Subject: [Freeswitch-users] how to record the conference manually? In-Reply-To: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> References: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> Message-ID: I suspect the conference is number at domain... do a conference list and check. /b On Jun 29, 2009, at 11:03 PM, jun yang wrote: > Hi,everyone. new to play with freeswitch. > i have setup freeswitch and can call each other very well. > now, a conference 3002 with several users in it. > i want to record 3002 manually, but can't get the way. > i have try fs_cli use the command: > conference 3002 record /tmp/foo.wav > it response: conference 3002 not found > any clue? > thanks in advance! From jason at jasonjgw.net Mon Jun 29 22:50:08 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 15:50:08 +1000 Subject: [Freeswitch-users] how to record the conference manually? In-Reply-To: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> References: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> Message-ID: <20090630055008.GA2693@jdc.jasonjgw.net> jun yang wrote: > now, a conference 3002 with several users in it. > i want to record 3002 manually, but can't get the way. > i have try fs_cli use the command: > conference 3002 record /tmp/foo.wav > it response: conference 3002 not found You need to specify the full conference name, including the domain. conference list will show you the full names of active conferences. From dome at tel.co.th Mon Jun 29 23:07:15 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 30 Jun 2009 13:07:15 +0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> Message-ID: <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> 2009/6/30 Michael Collins : > What kind of application are you building? Usually you want to use the > dialplan to initiate the call and then let the js do the logical heavy > lifting. I'm use js for callback solution. Dome C. > > -MC > > On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost wrote: >> >> Dear All, >> >> I try >> >> s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); >> if (s.ready()){ >> ? s.setVariable("nibble_rate", "2.5"); >> ? s.setVariable("nibble_account", "0838833133"); >> ? s.execute("nibblebill", "heartbeat 5"); >> ? bridge(session,s); >> }; >> >> my question is >> 1. How to cancel create s session (by dtmf ) like a * in bridge app >> 2. when i hangup before s session ready is posible to cancel ? >> >> Best Regards. >> >> Dome C. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Jun 29 23:17:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 23:17:42 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24261922.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <89F24CCA-53E0-4490-A901-C79C8806CBFA@freeswitch.org> <4A48F5D7.3050405@coppice.org> <26D74A59-E37D-4920-A1B3-0E83CE1FB811@freeswitch.org> <4A48FA71.3050406@coppice.org> <1246299408.3877.24.camel@dk-d820> <24261922.post@talk.nabble.com> Message-ID: <87f2f3b90906292317l31e7f64dkc22fd0e9b2b0bd8a@mail.gmail.com> On Mon, Jun 29, 2009 at 6:37 PM, Edmar Cruz wrote: > > Please give me some free drivers of G729... Please > This is the primary reason that most OSS guys hate G729 - it's patent encumbered BIG TIME. Check out this article on the main FS site: http://www.freeswitch.org/node/155 look at the patent claims on G729. G729A is even worse. Add to that the fact that the voice quality is poor and you have a recipe for a codec that telecom geeks don't really care for all that much. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/df16aa4e/attachment-0002.html From msc at freeswitch.org Mon Jun 29 23:19:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 29 Jun 2009 23:19:31 -0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> Message-ID: <87f2f3b90906292319v3d5dcee7ub2820097ff52f551@mail.gmail.com> can you post your script and dialplan? Let's take a look. -MC On Mon, Jun 29, 2009 at 11:07 PM, Dome Charoenyost wrote: > 2009/6/30 Michael Collins : > > What kind of application are you building? Usually you want to use the > > dialplan to initiate the call and then let the js do the logical heavy > > lifting. > I'm use js for callback solution. > > Dome C. > > > > > -MC > > > > On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost > wrote: > >> > >> Dear All, > >> > >> I try > >> > >> s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); > >> if (s.ready()){ > >> s.setVariable("nibble_rate", "2.5"); > >> s.setVariable("nibble_account", "0838833133"); > >> s.execute("nibblebill", "heartbeat 5"); > >> bridge(session,s); > >> }; > >> > >> my question is > >> 1. How to cancel create s session (by dtmf ) like a * in bridge app > >> 2. when i hangup before s session ready is posible to cancel ? > >> > >> Best Regards. > >> > >> Dome C. > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090629/0e3f2e46/attachment-0002.html From yj13535428332 at gmail.com Mon Jun 29 23:22:04 2009 From: yj13535428332 at gmail.com (jun yang) Date: Tue, 30 Jun 2009 14:22:04 +0800 Subject: [Freeswitch-users] how to record the conference manually? In-Reply-To: <20090630055008.GA2693@jdc.jasonjgw.net> References: <6536698d0906292103j37f1d393o5ae7efacad2161b9@mail.gmail.com> <20090630055008.GA2693@jdc.jasonjgw.net> Message-ID: <6536698d0906292322t54a63549q266f05e10b89c1be@mail.gmail.com> thanks,Brian West and Jason White. now it works fine. On Tue, Jun 30, 2009 at 1:50 PM, Jason White wrote: > jun yang wrote: > > now, a conference 3002 with several users in it. > > i want to record 3002 manually, but can't get the way. > > i have try fs_cli use the command: > > conference 3002 record /tmp/foo.wav > > it response: conference 3002 not found > > You need to specify the full conference name, including the domain. > > conference list > will show you the full names of active conferences. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/33eaf6f9/attachment-0002.html From darklion11 at yahoo.com Mon Jun 29 23:26:04 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 23:26:04 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> Message-ID: <24266762.post@talk.nabble.com> I need actually G729 license if there's any for freeswitch to call mobile phones... I already load it but has an issue on it passthrough mode... If i install mod_dandhi_codec it overwrites the existing G729 without license to a new G729 with license? dujinfang wrote: > >> > > Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/ > Linux/Windows, and I'd like to know some UAs supporting WB and UWB > codecs. > > Thanks. > >> Well said! Like Steve has pointed out in the past: G729/G729A is a >> race to the bottom. After using WB and UWB codecs all day every day >> for the past 6 months I just can't live with G729 or even GSM for >> that matter. However, to each his own. >> -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24266762.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brad.tuan at gmail.com Mon Jun 29 23:26:09 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 30 Jun 2009 14:26:09 +0800 Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? Message-ID: I have already set it in my FreeSwitch\conf\sip_profiles\external.xml: But my FS_A always return a 403 to FS_B, Where is the problem?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/433e7289/attachment-0002.html From darklion11 at yahoo.com Mon Jun 29 23:37:27 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 29 Jun 2009 23:37:27 -0700 (PDT) Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? In-Reply-To: References: Message-ID: <24266862.post@talk.nabble.com> Create a dialplan for FS A to communicate to FS B. dialplan/default/00_fsa.xml -> IP of FS B To check if your gateway succeed just type on sofia status check if gateway FreeSWITCH is there connection is set and ready to go :) Brad Tuan wrote: > > I have already set it in my FreeSwitch\conf\sip_profiles\external.xml: > > > > > > > > > > > > > But my FS_A always return a 403 to FS_B, Where is the problem?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-set-FS_A-as-a-gateway-of-FS_B---tp24266777p24266862.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yudha2008 at gmail.com Mon Jun 29 23:54:36 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 30 Jun 2009 12:24:36 +0530 Subject: [Freeswitch-users] javascript FIFO (First In First Out) In-Reply-To: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> References: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> Message-ID: *Hi Michael Jerris, Is there any other possible way to queue the inbound call in JavaScript. I am working on this process: step 1: I want the inbound call to be in queue through JavaScript step2: Then JavaScript will check most waiting agent and bridge the call to the most waiting agent ( this has been done in JavaScript itself) I want the incoming call to be in queue Is this possible through java script. Assist me to resolve this problem. Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/330a5a5b/attachment-0002.html From dome at tel.co.th Tue Jun 30 00:01:23 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 30 Jun 2009 14:01:23 +0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <87f2f3b90906292319v3d5dcee7ub2820097ff52f551@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> <87f2f3b90906291040m218269b1k77ef98098352cb79@mail.gmail.com> <8ccbff060906292307i34f4e4b3q7a4f233716f628f2@mail.gmail.com> <87f2f3b90906292319v3d5dcee7ub2820097ff52f551@mail.gmail.com> Message-ID: <8ccbff060906300001wef058c6m5edf3eafab9a57c0@mail.gmail.com> 2009/6/30 Michael Collins : > can you post your script and dialplan? Let's take a look. > -MC dialstr[i] is array like a sofia/external/1111 at xxx.xxx.xxx.xxx dial_option = "{absolute_codec_string='GSM,G729',ignore_early_media=false,originate_timeout=20,origination_caller_id=xxxxx} for (var i = 1; i <= 3; i++){ if (session.ready()){ session.execute("set", "hangup_after_bridge=false"); result = session.setAutoHangup(false); s1 = new Session(dial_option+dialstr[i]); } if (s1.ready()){ s1.setVariable("nibble_rate", "2.5"); s1.execute("set", "hangup_after_bridge=false"); s1.setVariable("nibble_account", acaller); s1.setVariable("provider_id", dialprovider_id[i]); s1.setVariable("provider", dialprovider[i]); s1.setVariable("service_charge", dialservice_charge[i]); s1.execute("set", "destination_number="+number); s1.execute("nibblebill", "heartbeat 5"); bridge(session,s1); console_log("notice", "Disconnect cause: " + s1.cause + " Code:"+s1.causecode+"\n"); }; if (s1.causecode==16 || s1.causecode==0){ i =10; }; }; > > On Mon, Jun 29, 2009 at 11:07 PM, Dome Charoenyost wrote: >> >> 2009/6/30 Michael Collins : >> > What kind of application are you building? Usually you want to use the >> > dialplan to initiate the call and then let the js do the logical heavy >> > lifting. >> I'm use js for callback solution. >> >> Dome C. >> >> > >> > -MC >> > >> > On Fri, Jun 26, 2009 at 11:45 PM, Dome Charoenyost >> > wrote: >> >> >> >> Dear All, >> >> >> >> I try >> >> >> >> s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); >> >> if (s.ready()){ >> >> ? s.setVariable("nibble_rate", "2.5"); >> >> ? s.setVariable("nibble_account", "0838833133"); >> >> ? s.execute("nibblebill", "heartbeat 5"); >> >> ? bridge(session,s); >> >> }; >> >> >> >> my question is >> >> 1. How to cancel create s session (by dtmf ) like a * in bridge app >> >> 2. when i hangup before s session ready is posible to cancel ? >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brad.tuan at gmail.com Tue Jun 30 00:05:28 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 30 Jun 2009 15:05:28 +0800 Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? Message-ID: I know that i need to set the dialplan, my problem is when FS_B send a REGISTER to FS_A, FS_A will return a 403 to FS_B Like this: 2009-06-30 15:03:25 [NOTICE] sofia_reg.c:305 sofia_reg_check_gateway() Registering FS_A 2009-06-30 15:03:25 [ERR] sofia_reg.c:1391 sofia_reg_handle_sip_r_register() FS_A Registration Failed with status Forbidden [403]. failure #2 2009-06-30 15:03:25 [WARNING] sofia_reg.c:334 sofia_reg_check_gateway() FS_A Failed Registration, setting retry to 30 seconds. > Create a dialplan for FS A to communicate to FS B. > > dialplan/default/00_fsa.xml > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > data="sofia/gateway/FreeSwitch/1$218.210.xxx.xxx"/> -> IP of FS B > > > To check if your gateway succeed > just type on sofia status check if gateway FreeSWITCH is there connection is > set and ready to go :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/38ca157f/attachment-0002.html From dujinfang at gmail.com Tue Jun 30 01:25:35 2009 From: dujinfang at gmail.com (seven) Date: Tue, 30 Jun 2009 16:25:35 +0800 Subject: [Freeswitch-users] javascript FIFO (First In First Out) In-Reply-To: References: <4BF1C3C8-DA43-4396-A28E-244D841D9FAF@jerris.com> Message-ID: <4BC016E6-75F0-48C6-B305-6D4C19799CF3@gmail.com> On Jun 30, 2009, at 2:54 PM, Baskar wrote: > Hi Michael Jerris, > > Is there any other possible way to queue the inbound call in > JavaScript. > > I am working on this process: > > step 1: I want the inbound call to be in queue through JavaScript > > step2: Then JavaScript will check most waiting agent and bridge the > call to the most waiting agent ( this has been done in JavaScript > itself) Just curious, why use a js while mod_fifo can do ? > > > I want the incoming call to be in queue Is this possible through > java script. > > Assist me to resolve this problem. > > > > Thanks with Regards, > N.Baskar > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/f03967c8/attachment-0002.html From saeedahmad1981 at gmail.com Tue Jun 30 02:04:30 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 30 Jun 2009 11:04:30 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: Hi, What is the best way to update to latest version if we are already running an older stable version? I am using SVN trunk, sources are in /usr/src and its installed in /usr/local/freeswitch >From the same src directory, is it possible to install latest rev in somewhere else (for example /opt/freeswitch), if everything is working ok then i replace it in /usr/local/freeswitch? or any other alternatives? Many thanks. On Tue, Jun 30, 2009 at 1:13 AM, Michael Collins wrote: > The FreeSWITCH development team is please to announce that there's a new > version of FreeSWITCH available. Please update as soon as you reasonable > can. More details available here: > http://www.freeswitch.org/node/195 > > We appreciate everyone's help in making FreeSWITCH better. Please keep > testing and reporting back! > > -Michael > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/9753e692/attachment-0002.html From fdelawarde at wirelessmundi.com Tue Jun 30 02:15:49 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 30 Jun 2009 11:15:49 +0200 Subject: [Freeswitch-users] Any advances on T.38 support for FS? Message-ID: <1246353349.30167.83.camel@luna.tc.commsmundi.com> Many issues on Asterisk's T.38 (or probably just on T.38?)... Could it convince those relying on this "modern" version of a 50yo technology to switch to and with FreeSwitch? Fran?ois. From jason at jasonjgw.net Tue Jun 30 02:14:56 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 19:14:56 +1000 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: <20090630091456.GA5371@jdc.jasonjgw.net> Saeed Ahmad wrote: > What is the best way to update to latest version if we are already running > an older stable version? You did ask for the best way, which is to build packages for your operating system, then use your operating system's package manager to install them and keep track of different versions. This way, you can be sure that the right files are installed, that old versions are cleanly deleted (unless there's an error in the post-installation script, in which case it's a bug) and you can use the package manager to find out what files are installed and where they reside. You can then upgrade or downgrade simply by installing a different version of the package. FreeSWITCH supports building Debian packages, and there is also support for Centos and Fedora. From yudha2008 at gmail.com Tue Jun 30 02:27:20 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 30 Jun 2009 14:57:20 +0530 Subject: [Freeswitch-users] Error in OpenZap Message-ID: *Hi, **I have installed the latest trunk with A102D Sangoma card when i load the openzap i get this error ** freeswitch at localhost.localdomain> load mod_openzap 2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules configured: 1 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '1' API CALL [load(mod_openzap)] output: -ERR [module load file routine returned an error] 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:1-15' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '1:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:17-31' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '2' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:1-15' freeswitch at localhost.localdomain> 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:17-31' 2009-06-30 14:47:33 [INFO] zap_io.c:2370 load_config() Configured 0 channel(s) 2009-06-30 14:47:33 [ERR] zap_io.c:2633 zap_global_init() No modules configured! 2009-06-30 14:47:33 [ERR] mod_openzap.c:2401 mod_openzap_load() Error loading OpenZAP 2009-06-30 14:47:33 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so **Module load routine returned an error** How can i resolve the above problem. how can i load the openzap My Configuration files default.xml /etc/wanpipe/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 0 PCIBUS = 2 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml /usr/local/freeswitch/libs/openzap/conf/openzap.conf [span wanpipe1] number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 **How can i resolve the above problem. how can i load the openzap. can some one assist me to resolve this process.* * -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/ca34bb7e/attachment-0002.html From raul at etellicom.com Tue Jun 30 02:37:41 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 30 Jun 2009 06:37:41 -0300 Subject: [Freeswitch-users] Error in OpenZap In-Reply-To: References: Message-ID: <1246354661.19445.13.camel@raul-laptop> Replace [span wanpipe1] with [span wanpipe 1] in your openzap.conf file - it's missing a space between wanpipe (the span type) and 1 (the span ID). Regards, Raul On Tue, 2009-06-30 at 14:57 +0530, Baskar wrote: > Hi, > > I have installed the latest trunk with A102D Sangoma card when i load > the openzap i get this error > > freeswitch at localhost.localdomain> load mod_openzap > 2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules > configured: 1 > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'name' / 'OpenZAP' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'number' / '1' > API CALL [load(mod_openzap)] output: > -ERR [module load file routine returned an error] > > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'trunk_type' / 'E1' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '1:1-15' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'd-channel' / '1:16' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '1:17-31' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'name' / 'OpenZAP' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'number' / '2' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'trunk_type' / 'E1' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '2:1-15' > freeswitch at localhost.localdomain> 2009-06-30 14:47:33 [ERR] > zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' > 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] > 'b-channel' / '2:17-31' > 2009-06-30 14:47:33 [INFO] zap_io.c:2370 load_config() Configured 0 > channel(s) > 2009-06-30 14:47:33 [ERR] zap_io.c:2633 zap_global_init() No modules > configured! > 2009-06-30 14:47:33 [ERR] mod_openzap.c:2401 mod_openzap_load() Error > loading OpenZAP > 2009-06-30 14:47:33 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading > module /usr/local/freeswitch/mod/mod_openzap.so > **Module load routine returned an error** > > How can i resolve the above problem. how can i load the openzap > > My Configuration files > > > default.xml > > > expression="^9(\d{5,15})$"> > > data="OpenZAP/1/a/${dialed_ext}"/> > > > > /etc/wanpipe/wanpipe1.conf > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 0 > PCIBUS = 2 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = CRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 120 > LBO = 120OH > TE_SIG_MODE = CCS > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 16 > TDMV_HW_DTMF = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_ECHO_OFF = NO > TDMV_HWEC = NO > > > /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml > > > > > > > > > > > > > > > > /usr/local/freeswitch/libs/openzap/conf/openzap.conf > > [span wanpipe1] > number => 1 > trunk_type => e1 > b-channel => 1:1-15 > d-channel => 1:16 > b-channel => 1:17-31 > > How can i resolve the above problem. how can i load the openzap. can > some one assist me to resolve this process. > > -- > Thanks with Regards, > N.Baskar > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saeedahmad1981 at gmail.com Tue Jun 30 02:56:23 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 30 Jun 2009 11:56:23 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <20090630091456.GA5371@jdc.jasonjgw.net> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <20090630091456.GA5371@jdc.jasonjgw.net> Message-ID: Can you give me more info with CentOS. I am more comfortable with SVN trunks, can i do the same with SVN trunks? Thanks On Tue, Jun 30, 2009 at 11:14 AM, Jason White wrote: > Saeed Ahmad wrote: > > What is the best way to update to latest version if we are already > running > > an older stable version? > > You did ask for the best way, which is to build packages for your operating > system, then use your operating system's package manager to install them > and > keep track of different versions. This way, you can be sure that the right > files are installed, that old versions are cleanly deleted (unless there's > an > error in the post-installation script, in which case it's a bug) and you > can > use the package manager to find out what files are installed and where they > reside. > > You can then upgrade or downgrade simply by installing a different version > of > the package. > > FreeSWITCH supports building Debian packages, and there is also support for > Centos and Fedora. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/2a0e1b12/attachment-0002.html From benke at inqnet.at Tue Jun 30 03:09:05 2009 From: benke at inqnet.at (Christian Benke) Date: Tue, 30 Jun 2009 12:09:05 +0200 Subject: [Freeswitch-users] Is there a group variable? Message-ID: <20090630120905.536234f4@plex> Hello! I have the following scenario: I want to check if a called extension is part of a group, or as an alternative, if it is a user in the directory. My basic intention is to find out if a 3-digit extension leads to a valid user - if it doesn't, some other action will happen. Maybe there are better ways to do this than to check for the parameters above, not sure though(Don't want to use a regexp for the available extensions though). If there is such a variable, it doesn't seem to be documented or i'm still too lost in the docs... Can you give me a pointer? Regards Christian From jason at jasonjgw.net Tue Jun 30 03:09:09 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 30 Jun 2009 20:09:09 +1000 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <20090630091456.GA5371@jdc.jasonjgw.net> Message-ID: <20090630100909.GA18070@jdc.jasonjgw.net> Saeed Ahmad wrote: > Can you give me more info with CentOS. > > I am more comfortable with SVN trunks, can i do the same with SVN trunks? Yes. There is a spec file in the source tree for building packages. There should be instructions on the wiki explaining how to use it - if not, someone who is more familiar with rpm-based distributions than I am will be able to refer you to instructions on how to build an rpm package. From helmut.kuper at ewetel.de Tue Jun 30 04:29:17 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 30 Jun 2009 13:29:17 +0200 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <019601c9f92b$036c02d0$0a440870$@net> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> Message-ID: <4A49F70D.1080904@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Drago, sorry for correcting you, but Snom's touch scrren phone is the 870 ... http://www.snom.com/en/products/snom-870/ cheers Helmut On 30.06.2009 04:32, Drago Totev wrote: > Actually, Snom does have a version with color LCD touch screen - model 820. > I'm not sure if it is in mass production yet. > > Regarding the "nice looking"... suit yourself :-) : > http://www.snom.com/en/products/snom-820/ > > Drago -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKSfcN4tZeNddg3dwRAlKBAJ9dDp51Ki2CmY+VxEc/YvO5++xIIACfVca8 Dpc6N3A/9cPai8ujkSosv/A= =bCzJ -----END PGP SIGNATURE----- From raffaele.p.guidi at gmail.com Tue Jun 30 04:46:34 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 30 Jun 2009 13:46:34 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: Hi, I would like to give a try to this and all other "pre" releases but being tied to windows platforms (and not having a C compiler available) I would need an MSI installer. Is there a way to add a windows build to the (pre)release process (without doubling your work, of course ;)? I think one of the selling points of FreeSWITCH is the ability to seamlessly run on windows - something that asterisk cannot even dream of, and that yate promises but fails to completely fulfill. Regards, Raffaele On Tue, Jun 30, 2009 at 01:13, Michael Collins wrote: > The FreeSWITCH development team is please to announce that there's a new > version of FreeSWITCH available. Please update as soon as you reasonable > can. More details available here: > http://www.freeswitch.org/node/195 > > We appreciate everyone's help in making FreeSWITCH better. Please keep > testing and reporting back! > > -Michael > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/6f09a98a/attachment-0002.html From jay.fenton at howlertech.com Tue Jun 30 00:03:34 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Tue, 30 Jun 2009 08:03:34 +0100 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24266762.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> Message-ID: Hi Edmar, > I need actually G729 license if there's any for freeswitch to call > mobile > phones... I already load it but has an issue on it passthrough > mode... If i > install mod_dandhi_codec it overwrites the existing G729 without > license to > a new G729 with license? Are you sure your mobile phones support G729? Most phones have hardware acceleration for AMR-NB and not G729, so most of the VoIP-over-WiFi providers use that codec instead. I believe G729 would have to be implemented in software only (without hardware assistance) and would drain the battery fairly quickly as a result. As other have pointed out on the list, and just in case you missed it mod_dahdi_codec is for use with a PCI accelerator card (the TC400B) that Digium (http://www.digium.com/) sell and will not work without it. With that card you can get up to 120 concurrent G729A-G711 calls. These days you can get much better performance from pure software codecs running on Intel/AMD systems than using the above card. We (Howler) happen to have announced a G729A codec for FreeSWITCH a few days ago - you can get more info here: http://www.howlertech.com/products/howlets/ Hope that helps. -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From jay.fenton at howlertech.com Tue Jun 30 00:11:11 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Tue, 30 Jun 2009 08:11:11 +0100 Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729? In-Reply-To: <8cc991dd0906292020u3528f483v1462e049877c664d@mail.gmail.com> References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com> <20090630014115.GA23733@jdc.jasonjgw.net> <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org> <8cc991dd0906291858g44b9fc0p21a8eab45427c422@mail.gmail.com> <019601c9f92b$036c02d0$0a440870$@net> <1246330532.4148.3.camel@sodium> <8cc991dd0906292020u3528f483v1462e049877c664d@mail.gmail.com> Message-ID: <9D287080-0FDE-4DD6-871D-0D00DA132438@howlertech.com> > Well the 8XX series is much prettier than the 3XX series. Though I > wonder if anyone has thought of porting FreeSWITCH to the Palm Pre? Palm Pre is Linux-based, so shouldn't be that difficult so long as you can get a working portaudio up and running (it looks like pulseaudio has already been ported, so I presume ALSA is ready and waiting in the kernel - FS might just build out of the box!). -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From lubimov at neolant.ru Tue Jun 30 07:21:06 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Tue, 30 Jun 2009 18:21:06 +0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? Message-ID: <4A4A1F52.2000307@neolant.ru> I sofia_reg.c:1765 have two user records - good #110 and bad #111. bad.xml: good.xml: Good user 110 work without any problem. But user "Bad" user 111 can't register to freeswitch. In log I can see only one message - 2009-06-30 16:31:36.590970 [WARNING] sofia_reg.c:1765 Cant register a pointer. good user exists: freeswitch at internal> user_exists id 110 neolant.ru true and bad user exists! freeswitch at internal> user_exists id 111 neolant.ru true good user don't have attr type: freeswitch at internal> user_data 110 at neolant.ru attr type -ERR no reply but bad user have attr type! : freeswitch at internal> user_data 111 at neolant.ru attr type pointer Good user have password: freeswitch at internal> user_data 110 at neolant.ru param password 123456 But bad user no have param pasword! freeswitch at internal> user_data 111 at neolant.ru param password -ERR no reply What's wrong in these configuration? How I can debug and resolve these problems? From msc at freeswitch.org Tue Jun 30 08:02:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 30 Jun 2009 08:02:55 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: <8DE8B0BC-0519-426A-A8A1-99ABDA65B930@freeswitch.org> If you're already on trunk then just do "make current" -MC Sent from my iPhone On Jun 30, 2009, at 2:04 AM, Saeed Ahmad wrote: > Hi, > > What is the best way to update to latest version if we are already > running an older stable version? > > I am using SVN trunk, sources are in /usr/src and its installed in / > usr/local/freeswitch > > >From the same src directory, is it possible to install latest rev > in somewhere else (for example /opt/freeswitch), if everything is > working ok then i replace it in /usr/local/freeswitch? > > or any other alternatives? > > Many thanks. > > > > On Tue, Jun 30, 2009 at 1:13 AM, Michael Collins > wrote: > The FreeSWITCH development team is please to announce that there's a > new version of FreeSWITCH available. Please update as soon as you > reasonable can. More details available here: > http://www.freeswitch.org/node/195 > > We appreciate everyone's help in making FreeSWITCH better. Please > keep testing and reporting back! > > -Michael > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/e36bb83c/attachment-0002.html From carlos.talbot at gmail.com Tue Jun 30 08:29:58 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Tue, 30 Jun 2009 10:29:58 -0500 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> Message-ID: <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> I do make an effort to update the svn MSI every time a new release is announced. The current MSI was posted this morning (svn 14043) and should be synced up by this evening (CST). Carlos http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries On Tue, Jun 30, 2009 at 6:46 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > Hi, I would like to give a try to this and all other "pre" releases but > being tied to windows platforms (and not having a C compiler available) I > would need an MSI installer. Is there a way to add a windows build to the > (pre)release process (without doubling your work, of course ;)? I think one > of the selling points of FreeSWITCH is the ability to seamlessly run on > windows - something that asterisk cannot even dream of, and that yate > promises but fails to completely fulfill. > Regards, > Raffaele > > On Tue, Jun 30, 2009 at 01:13, Michael Collins wrote: > >> The FreeSWITCH development team is please to announce that there's a new >> version of FreeSWITCH available. Please update as soon as you reasonable >> can. More details available here: >> http://www.freeswitch.org/node/195 >> >> We appreciate everyone's help in making FreeSWITCH better. Please keep >> testing and reporting back! >> >> -Michael >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/1291e3a4/attachment-0002.html From mike at jerris.com Tue Jun 30 08:49:30 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 11:49:30 -0400 Subject: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h In-Reply-To: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> References: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> Message-ID: <5CE0CED3-A08D-4D18-826C-7C64E8C7E1BB@jerris.com> I think that detection is not working on mac right due to it looking in default search paths. I am in process of fixing this to use in tree libtiff soon so this should fix this issue. Mike On Jun 29, 2009, at 11:08 PM, seven wrote: > Hi, > > I'm on the latest svn 14041, and I have tiff installed with "port > install tiff", how can I tell FS to find libtiff at /opt/local/ > include? > > on Linux the package should be libtiff-dev of libtiff-devel, but I > think tiff is the equivalent on Mac. > > previous versions build ok on my Mac, possible to include the libtiff > in trunk like other libs? > > checking tiffio.h usability... no > checking tiffio.h presence... no > checking for tiffio.h... no > checking pthread.h usability... yes > checking pthread.h presence... yes > checking for pthread.h... yes > checking X11/X.h usability... yes > checking X11/X.h presence... yes > checking for X11/X.h... yes > checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h > usability... yes > checking libxml/xmlmemory.h presence... yes > checking for libxml/xmlmemory.h... yes > checking libxml/parser.h usability... yes > checking libxml/parser.h presence... yes > checking for libxml/parser.h... yes > checking libxml/xinclude.h usability... yes > checking libxml/xinclude.h presence... yes > checking for libxml/xinclude.h... yes > checking FL/Fl.H usability... no > checking FL/Fl.H presence... no > checking for FL/Fl.H... no > checking FL/Fl_Overlay_Window.H usability... no > checking FL/Fl_Overlay_Window.H presence... no > checking for FL/Fl_Overlay_Window.H... no > checking FL/Fl_Light_Button.H usability... no > checking FL/Fl_Light_Button.H presence... no > checking for FL/Fl_Light_Button.H... no > checking FL/fl_draw.H usability... no > checking FL/fl_draw.H presence... no > checking for FL/fl_draw.H... no > checking FL/Fl_Cartesian.H usability... no > checking FL/Fl_Cartesian.H presence... no > checking for FL/Fl_Cartesian.H... no > checking FL/Fl_Audio_Meter.H usability... no > checking FL/Fl_Audio_Meter.H presence... no > checking for FL/Fl_Audio_Meter.H... no > checking for cos in -lm... yes > checking for library containing sinf... none required > checking for library containing cosf... none required > checking for library containing tanf... none required > checking for library containing asinf... none required > checking for library containing acosf... none required > checking for library containing atanf... none required > checking for library containing atan2f... none required > checking for library containing ceilf... none required > checking for library containing floorf... none required > checking for library containing powf... none required > checking for library containing expf... none required > checking for library containing logf... none required > checking for library containing log10f... none required > checking for TIFFOpen in -ltiff... no > configure: error: "Can't build without libtiff (does your system > require a libtiff-devel package?)" > configure: error: ./configure.gnu failed for libs/spandsp > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 30 09:03:49 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:03:49 -0400 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Message-ID: the bridge app already does all this for you doesn't it (along with bind_meta) ? Mike On Jun 27, 2009, at 2:45 AM, Dome Charoenyost wrote: > Dear All, > > I try > > s = new Session("sofia/external/xxx at xxx.xxx.xxx.xxx); > if (s.ready()){ > s.setVariable("nibble_rate", "2.5"); > s.setVariable("nibble_account", "0838833133"); > s.execute("nibblebill", "heartbeat 5"); > bridge(session,s); > }; > > my question is > 1. How to cancel create s session (by dtmf ) like a * in bridge app > 2. when i hangup before s session ready is posible to cancel ? > > Best Regards. > > Dome C. From dujinfang at gmail.com Tue Jun 30 09:04:48 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 1 Jul 2009 00:04:48 +0800 Subject: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h In-Reply-To: <5CE0CED3-A08D-4D18-826C-7C64E8C7E1BB@jerris.com> References: <8C787EE4-7E0B-4D5C-86BE-9135D0D62D85@gmail.com> <5CE0CED3-A08D-4D18-826C-7C64E8C7E1BB@jerris.com> Message-ID: Thanks. I installed libtiff from source to /usr/local/ and now it works. On Jun 30, 2009, at 11:49 PM, Michael Jerris wrote: > I think that detection is not working on mac right due to it looking > in default search paths. I am in process of fixing this to use in > tree libtiff soon so this should fix this issue. > > Mike > > On Jun 29, 2009, at 11:08 PM, seven wrote: > >> Hi, >> >> I'm on the latest svn 14041, and I have tiff installed with "port >> install tiff", how can I tell FS to find libtiff at /opt/local/ >> include? >> >> on Linux the package should be libtiff-dev of libtiff-devel, but I >> think tiff is the equivalent on Mac. >> >> previous versions build ok on my Mac, possible to include the libtiff >> in trunk like other libs? >> >> checking tiffio.h usability... no >> checking tiffio.h presence... no >> checking for tiffio.h... no >> checking pthread.h usability... yes >> checking pthread.h presence... yes >> checking for pthread.h... yes >> checking X11/X.h usability... yes >> checking X11/X.h presence... yes >> checking for X11/X.h... yes >> checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h >> usability... yes >> checking libxml/xmlmemory.h presence... yes >> checking for libxml/xmlmemory.h... yes >> checking libxml/parser.h usability... yes >> checking libxml/parser.h presence... yes >> checking for libxml/parser.h... yes >> checking libxml/xinclude.h usability... yes >> checking libxml/xinclude.h presence... yes >> checking for libxml/xinclude.h... yes >> checking FL/Fl.H usability... no >> checking FL/Fl.H presence... no >> checking for FL/Fl.H... no >> checking FL/Fl_Overlay_Window.H usability... no >> checking FL/Fl_Overlay_Window.H presence... no >> checking for FL/Fl_Overlay_Window.H... no >> checking FL/Fl_Light_Button.H usability... no >> checking FL/Fl_Light_Button.H presence... no >> checking for FL/Fl_Light_Button.H... no >> checking FL/fl_draw.H usability... no >> checking FL/fl_draw.H presence... no >> checking for FL/fl_draw.H... no >> checking FL/Fl_Cartesian.H usability... no >> checking FL/Fl_Cartesian.H presence... no >> checking for FL/Fl_Cartesian.H... no >> checking FL/Fl_Audio_Meter.H usability... no >> checking FL/Fl_Audio_Meter.H presence... no >> checking for FL/Fl_Audio_Meter.H... no >> checking for cos in -lm... yes >> checking for library containing sinf... none required >> checking for library containing cosf... none required >> checking for library containing tanf... none required >> checking for library containing asinf... none required >> checking for library containing acosf... none required >> checking for library containing atanf... none required >> checking for library containing atan2f... none required >> checking for library containing ceilf... none required >> checking for library containing floorf... none required >> checking for library containing powf... none required >> checking for library containing expf... none required >> checking for library containing logf... none required >> checking for library containing log10f... none required >> checking for TIFFOpen in -ltiff... no >> configure: error: "Can't build without libtiff (does your system >> require a libtiff-devel package?)" >> configure: error: ./configure.gnu failed for libs/spandsp >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Jun 30 09:05:28 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:05:28 -0400 Subject: [Freeswitch-users] How to set FS_A as a gateway of FS_B?? In-Reply-To: References: Message-ID: <2F02DCD0-71EF-4787-9BE6-D4D667E53132@jerris.com> If you don't need authentication, you don't need a gateway, if you do, you will need to setup a user on the other box to register to. On Jun 30, 2009, at 3:05 AM, Brad Tuan wrote: > I know that i need to set the dialplan, > > my problem is when FS_B send a REGISTER to FS_A, FS_A will return a > 403 to FS_B > > Like this: > > 2009-06-30 15:03:25 [NOTICE] sofia_reg.c:305 > sofia_reg_check_gateway() Registering FS_A > 2009-06-30 15:03:25 [ERR] sofia_reg.c:1391 > sofia_reg_handle_sip_r_register() > FS_A Registration Failed with status Forbidden > [403]. failure #2 > 2009-06-30 15:03:25 [WARNING] sofia_reg.c:334 > sofia_reg_check_gateway() > FS_A Failed Registration, setting retry to 30 > seconds. > > > Create a dialplan for FS A to communicate to FS B. > > > > dialplan/default/00_fsa.xml > > > > > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > > > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > > > data="sofia/gateway/FreeSwitch/1$218.210.xxx.xxx"/> -> IP of > FS B > > > > > > > To check if your gateway succeed > > > just type on sofia status check if gateway FreeSWITCH is there > connection is > > set and ready to go :) From mike at jerris.com Tue Jun 30 09:10:20 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:10:20 -0400 Subject: [Freeswitch-users] Is there a group variable? In-Reply-To: <20090630120905.536234f4@plex> References: <20090630120905.536234f4@plex> Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#in_group http://wiki.freeswitch.org/wiki/Mod_commands#user_exists On Jun 30, 2009, at 6:09 AM, Christian Benke wrote: > Hello! > > I have the following scenario: > > I want to check if a called extension is part of a group, or as an > alternative, if it is a user in the directory. > My basic intention is to find out if a 3-digit extension leads to a > valid user - if it doesn't, some other action will happen. > Maybe there are better ways to do this than to check for the > parameters > above, not sure though(Don't want to use a regexp for the available > extensions though). > > If there is such a variable, it doesn't seem to be documented or i'm > still too lost in the docs... Can you give me a pointer? From mike at jerris.com Tue Jun 30 09:12:26 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:12:26 -0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? In-Reply-To: <4A4A1F52.2000307@neolant.ru> References: <4A4A1F52.2000307@neolant.ru> Message-ID: you have a pointer somewhere in your directory for that user, hard to see without seeing the whole config, but grep for 111 and see what else you find. Mike On Jun 30, 2009, at 10:21 AM, Alexey Lubimov wrote: > I sofia_reg.c:1765 have two user records - good #110 and bad #111. > > bad.xml: > > > > > > > > > value="domestic,international,local"/> > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > good.xml: > > > > > > > > > value="domestic,international,local"/> > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > Good user 110 work without any problem. But user "Bad" user 111 can't > register to freeswitch. > > In log I can see only one message - 2009-06-30 16:31:36.590970 > [WARNING] sofia_reg.c:1765 Cant register a pointer. > > good user exists: > freeswitch at internal> user_exists id 110 neolant.ru > true > > and bad user exists! > freeswitch at internal> user_exists id 111 neolant.ru > true > > good user don't have attr type: > > freeswitch at internal> user_data 110 at neolant.ru attr type > -ERR no reply > > but bad user have attr type! : > > freeswitch at internal> user_data 111 at neolant.ru attr type > pointer > > > Good user have password: > > freeswitch at internal> user_data 110 at neolant.ru param password > 123456 > > But bad user no have param pasword! > > freeswitch at internal> user_data 111 at neolant.ru param password > -ERR no reply > > > What's wrong in these configuration? How I can debug and resolve these > problems? From mike at jerris.com Tue Jun 30 09:07:13 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 30 Jun 2009 12:07:13 -0400 Subject: [Freeswitch-users] Any advances on T.38 support for FS? In-Reply-To: <1246353349.30167.83.camel@luna.tc.commsmundi.com> References: <1246353349.30167.83.camel@luna.tc.commsmundi.com> Message-ID: <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> We currently support t.38 passthrough only using proxy_media mode. T. 38 gateway is on the roadmap but not yet close to complete. Mike On Jun 30, 2009, at 5:15 AM, Fran?ois Delawarde wrote: > Many issues on Asterisk's T.38 (or probably just on T.38?)... > > Could it convince those relying on this "modern" version of a 50yo > technology to switch to and with FreeSwitch? From msc at freeswitch.org Tue Jun 30 09:50:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 30 Jun 2009 09:50:29 -0700 Subject: [Freeswitch-users] How to cancel session in Javascript In-Reply-To: References: <8ccbff060906262345y28c89049r96b126f109d580ff@mail.gmail.com> Message-ID: <87f2f3b90906300950m6514972dxa26c43bfbb9785ce@mail.gmail.com> On Tue, Jun 30, 2009 at 9:03 AM, Michael Jerris wrote: > the bridge app already does all this for you doesn't it (along with > bind_meta) ? > > Mike > In other words, everything you want is available in the dialplan with no overheard from launching a JS. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/cf260881/attachment-0002.html From raffaele.p.guidi at gmail.com Tue Jun 30 10:45:29 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 30 Jun 2009 19:45:29 +0200 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> Message-ID: Ok, so tomorrow I'll find http://files.freeswitch.org/freeswitch-1.0.4pre9.msi? Thanks a lot, Raffaele On Tue, Jun 30, 2009 at 17:29, Carlos Talbot wrote: > I do make an effort to update the svn MSI every time a new release is > announced. The current MSI was posted this morning (svn 14043) and should be > synced up by this evening (CST). > Carlos > > > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries > > > On Tue, Jun 30, 2009 at 6:46 AM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > >> Hi, I would like to give a try to this and all other "pre" releases but >> being tied to windows platforms (and not having a C compiler available) I >> would need an MSI installer. Is there a way to add a windows build to the >> (pre)release process (without doubling your work, of course ;)? I think one >> of the selling points of FreeSWITCH is the ability to seamlessly run on >> windows - something that asterisk cannot even dream of, and that yate >> promises but fails to completely fulfill. >> Regards, >> Raffaele >> >> On Tue, Jun 30, 2009 at 01:13, Michael Collins wrote: >> >>> The FreeSWITCH development team is please to announce that there's a new >>> version of FreeSWITCH available. Please update as soon as you reasonable >>> can. More details available here: >>> http://www.freeswitch.org/node/195 >>> >>> We appreciate everyone's help in making FreeSWITCH better. Please keep >>> testing and reporting back! >>> >>> -Michael >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/66bc7fc1/attachment-0002.html From brian at freeswitch.org Tue Jun 30 11:01:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 30 Jun 2009 13:01:32 -0500 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9 In-Reply-To: References: <87f2f3b90906291613x7101ecb5t48ad458ee969c760@mail.gmail.com> <5800526b0906300829g535328bes7a2504ed7c52f472@mail.gmail.com> Message-ID: <223FE753-8F58-4196-B9A1-8F9BE8D08067@freeswitch.org> Maybe! We might hold it hostage! ;) muahhahaha /b On Jun 30, 2009, at 12:45 PM, Raffaele P. Guidi wrote: > Ok, so tomorrow I'll find http://files.freeswitch.org/freeswitch-1.0.4pre9.msi? > > Thanks a lot, > Raffaele -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/6d8d446d/attachment-0002.html From marketing at cluecon.com Tue Jun 30 14:58:50 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 30 Jun 2009 14:58:50 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Great News: Early Bird Extended Message-ID: <87f2f3b90906301458n584ea8e8ta848cf1f184a8976@mail.gmail.com> Great news for all of those who've not been able to sign up yet for ClueCon: we've decided to extend the early bird special through Tuesday July 21! Note: this is also the last day you can book a room with the Wyndham, but please don't wait until the last minute. More good news and reasons to sign up right away: fun giveaways! Our sponsors and partners are supplying goodies to give away to conference attendees. Here are some highlights: >From Sangoma - B600 cards (qty 3) and A101D (qty 1) >From Snom - 20 phones, mix of models 320, 360, 270 Stay tuned for more updates and surprises about ClueCon 2009! -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/f80f31ee/attachment-0002.html From darklion11 at yahoo.com Tue Jun 30 18:46:01 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 30 Jun 2009 18:46:01 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> Message-ID: <24282895.post@talk.nabble.com> So what codec supports mobile phones? jfenton wrote: > > > Hi Edmar, > >> I need actually G729 license if there's any for freeswitch to call >> mobile >> phones... I already load it but has an issue on it passthrough >> mode... If i >> install mod_dandhi_codec it overwrites the existing G729 without >> license to >> a new G729 with license? > > Are you sure your mobile phones support G729? Most phones have hardware > acceleration for AMR-NB and not G729, so most of the VoIP-over-WiFi > providers use that codec instead. I believe G729 would have to be > implemented in software only (without hardware assistance) and would > drain the battery fairly quickly as a result. > > As other have pointed out on the list, and just in case you missed it > mod_dahdi_codec is for use with a PCI accelerator card (the TC400B) > that Digium (http://www.digium.com/) sell and will not work without > it. With that card you can get up to 120 concurrent G729A-G711 calls. > > These days you can get much better performance from pure software > codecs running on Intel/AMD systems than using the above card. We > (Howler) happen to have announced a G729A codec for FreeSWITCH a > few days ago - you can get more info here: > > http://www.howlertech.com/products/howlets/ > > Hope that helps. > > -- > Regards, > > Jay Fenton, CTO > Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ > tel: +44 207 099 7095 fax: +44 207 099 7098 > http://www.howlertech.com/ > http://www.linkedin.com/in/jfenton > > Registered in England & Wales, Company No. 06285634 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24282895.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From craig at overthewire.com.au Tue Jun 30 19:03:38 2009 From: craig at overthewire.com.au (Craig Askings) Date: Wed, 1 Jul 2009 12:03:38 +1000 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24282895.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <8cc991dd0906301903k2dd15de9tdd9b7b9a77338ce1@mail.gmail.com> GSM 2009/7/1 Edmar Cruz : > > So what codec supports mobile phones? > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From steveu at coppice.org Tue Jun 30 20:05:20 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 01 Jul 2009 11:05:20 +0800 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24282895.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> Message-ID: <4A4AD270.30204@coppice.org> Edmar Cruz wrote: > So what codec supports mobile phones? > The main codecs used by mobile phones are: GSM FR The original GSM code, largely replaced by later codecs (some VoIP stuff uses this) GSM HR The half rate codec for GSM GSM EFR A later improved full rate codec that largely replaced GSM FR on GSM networks AMR-NB The current start of the art narrowband codec for GSM and UMTS networks AMR-WB The wide band codec for UMTS, though most networks seem to block it EVRC The main codec for the CDMA networks Each of these codecs sounds pretty respectable, and G.729 sounds pretty respectable. However, if you transcode from one to another the result can be pretty bad. Steve From rupa at rupa.com Tue Jun 30 20:22:27 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 30 Jun 2009 22:22:27 -0500 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A4AD270.30204@coppice.org> References: <24251951.post@talk.nabble.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <4A4AD270.30204@coppice.org> Message-ID: And unless you are directly connecting to the cell phone provider you are going to be converted to ULAW/ALAW to traverse the PSTN. So there is no advantage going to a native GSM codec only to have it expanded out to G711 and then back to GSM codec. On Tue, Jun 30, 2009 at 10:05 PM, Steve Underwood wrote: > Edmar Cruz wrote: > > So what codec supports mobile phones? > > > The main codecs used by mobile phones are: > GSM FR The original GSM code, largely replaced by later > codecs (some VoIP stuff uses this) > GSM HR The half rate codec for GSM > GSM EFR A later improved full rate codec that largely replaced > GSM FR on GSM networks > AMR-NB The current start of the art narrowband codec for GSM > and UMTS networks > AMR-WB The wide band codec for UMTS, though most networks > seem to block it > EVRC The main codec for the CDMA networks > > Each of these codecs sounds pretty respectable, and G.729 sounds pretty > respectable. However, if you transcode from one to another the result > can be pretty bad. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/33157112/attachment-0002.html From darklion11 at yahoo.com Tue Jun 30 20:43:34 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Tue, 30 Jun 2009 20:43:34 -0700 (PDT) Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <4A4AD270.30204@coppice.org> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <4A4AD270.30204@coppice.org> Message-ID: <24283663.post@talk.nabble.com> I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i shall load to G729 for freeswitch that needs a license? Steve Underwood wrote: > > Edmar Cruz wrote: >> So what codec supports mobile phones? >> > The main codecs used by mobile phones are: > GSM FR The original GSM code, largely replaced by later > codecs (some VoIP stuff uses this) > GSM HR The half rate codec for GSM > GSM EFR A later improved full rate codec that largely replaced > GSM FR on GSM networks > AMR-NB The current start of the art narrowband codec for GSM > and UMTS networks > AMR-WB The wide band codec for UMTS, though most networks > seem to block it > EVRC The main codec for the CDMA networks > > Each of these codecs sounds pretty respectable, and G.729 sounds pretty > respectable. However, if you transcode from one to another the result > can be pretty bad. > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24283663.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mythicalbox at weavver.com Tue Jun 30 21:11:36 2009 From: mythicalbox at weavver.com (Mitchel Constantin) Date: Tue, 30 Jun 2009 21:11:36 -0700 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? Message-ID: Hello, I'm experiencing a bug that I've been working on most of today. I can not call between two SIP phones that register successfully. In order to diagnose it, I have removed my FreeSWITCH server out of the NAT/firewall to try and eliminate any such issues with these things. Here is how I ran into the issue: 1. Started from sample configs 2. Enabled xml_curl and wrote the associated script to generate XML and had the phones authenticating but was forced to use the IP address of the server as the domain in my directory XML. 3. I tested calls and they worked using this syntax: originate sofia/internal/mythicalbox%205.134.225.20 (the server's ip) sofia/internal/johndoe%205.134.225.20 4. Next to remove the "limit" on using only the IP as the domain for users, I commented out force-register-domain and force-register-db-domain in internal.xml. 5. My phones now register using the correct domain name (i.e. weavver.com) instead of the IP address (205.134.225.20) as the domain. 6. Now the problem... My originate command no longer works using the new syntax: originate sofia/internal/mythicalbox%weavver.comsofia/internal/johndoe% weavver.com The phones do show up as registered when I type "sofia status profile internal": API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 205.134.225.20 SIP-IP 205.134.225.20 URL sip:mod_sofia at 205.134.225.20:5060 BIND-URL sip:mod_sofia at 205.134.225.20:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 9 FAILED-CALLS-IN 3 CALLS-OUT 8 FAILED-CALLS-OUT 18 Registrations: ================================================================================================= Call-ID: ZWQ3NjRhZWI3MDc4ZDhjNTdhZDU2ZGVkM2JmYjg3NTc. User: mythicalbox at weavver.com Contact: "mythicalbox" Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-06-30 22:43:47) Host: duck.weavver.com IP: 64.183.110.250 Port: 65438 Auth-User: mythicalbox Auth-Realm: weavver.com Call-ID: Y2E0NWZiZWZjOTYxYjVhNmQ5ZDkyMzJjMzUxYWM0ZGM. User: johndoe at weavver.com Contact: "johndoe" Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-06-30 22:47:37) Host: duck.weavver.com IP: 64.183.110.250 Port: 65512 Auth-User: johndoe Auth-Realm: weavver.com ================================================================================================= FYI, FreeSWITCH is on a public IP address on the 'nets and the phones are behind the same firewall on a different public IP address on the internet. Thank you in advance for any help! :) -- Mitchel Constantin Weavver, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090630/43fe93c4/attachment-0002.html From msc at freeswitch.org Tue Jun 30 21:40:12 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 30 Jun 2009 21:40:12 -0700 Subject: [Freeswitch-users] Is there any license G729? In-Reply-To: <24283663.post@talk.nabble.com> References: <24251951.post@talk.nabble.com> <20090629105113.GA4756@jdc.jasonjgw.net> <24252099.post@talk.nabble.com> <87f2f3b90906290817p378d208cra3350241e440e2e8@mail.gmail.com> <6b65470d0906290842y107909d8k5f64a9e20099c157@mail.gmail.com> <9618647f0906290848r197612b0h6c7f00799aed4920@mail.gmail.com> <4A48E757.60103@coppice.org> <87f2f3b90906290952q72da5aefr9af336f30c5fc7e0@mail.gmail.com> <24266762.post@talk.nabble.com> <24282895.post@talk.nabble.com> <4A4AD270.30204@coppice.org> <24283663.post@talk.nabble.com> Message-ID: <39D35AD1-05A2-4D42-AEF1-09216777B66C@freeswitch.org> On Jun 30, 2009, at 8:43 PM, Edmar Cruz wrote: > > I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i > shall load > to G729 for freeswitch that needs a license? > You need a license if you are transcoding to or from G729. If you are just passing the media stream or you stay out of the media stream then you do not need a license. -MC > Steve Underwood wrote: >> >> Edmar Cruz wrote: >>> So what codec supports mobile phones? >>> >> The main codecs used by mobile phones are: >> GSM FR The original GSM code, largely replaced by later >> codecs (some VoIP stuff uses this) >> GSM HR The half rate codec for GSM >> GSM EFR A later improved full rate codec that largely >> replaced >> GSM FR on GSM networks >> AMR-NB The current start of the art narrowband codec for >> GSM >> and UMTS networks >> AMR-WB The wide band codec for UMTS, though most networks >> seem to block it >> EVRC The main codec for the CDMA networks >> >> Each of these codecs sounds pretty respectable, and G.729 sounds >> pretty >> respectable. However, if you transcode from one to another the result >> can be pretty bad. >> >> Steve >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24283663.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yudha2008 at gmail.com Tue Jun 30 23:29:14 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 1 Jul 2009 11:59:14 +0530 Subject: [Freeswitch-users] Error in OpenZap In-Reply-To: <1246354661.19445.13.camel@raul-laptop> References: <1246354661.19445.13.camel@raul-laptop> Message-ID: *Hi, i have changed the openzap.conf file but still i get the same error **[span wanpipe 1] number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31* * freeswitch at localhost.localdomain> load mod_libpri API CALL [load(mod_libpri)] output: -ERR [module load file routine returned an error] 2009-07-01 11:27:51 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_libpri.so **/usr/local/freeswitch/mod/mod_libpri.so: cannot open shared object file: No such file or directory** freeswitch at localhost.localdomain> load mod_openzap 2009-07-01 11:28:04 [NOTICE] zap_io.c:2626 zap_global_init() Modules configured: 1 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '1' API CALL [load(mod_openzap)] output: -ERR [module load file routine returned an error] 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:1-15' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '1:16' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:17-31' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '2' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:1-15' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:17-31' 2009-07-01 11:28:04 [INFO] zap_io.c:2370 load_config() Configured 0 channel(s) 2009-07-01 11:28:04 [ERR] zap_io.c:2633 zap_global_init() No modules configured! 2009-07-01 11:28:04 [ERR] mod_openzap.c:2401 mod_openzap_load() Error loading OpenZAP 2009-07-01 11:28:04 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so **Module load routine returned an error** * *-- Thanks with Regards, N.Baskar* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/bcc735c4/attachment-0002.html From god.nirvana at gmail.com Tue Jun 30 23:48:03 2009 From: god.nirvana at gmail.com (qian ma) Date: Wed, 1 Jul 2009 14:48:03 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? Message-ID: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> hi all freeswitch support PCMU only? i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but freeswitch still support PCMU only, below is the trace: 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9876 at 58.212.219.104 [KILL] 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal sofia/maq/9876 at 58.212.219.104 [BREAK] 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( sofia/maq/9876 at 58.212.219.104) State HANGUP how to configure the freeswitch?? support more codecs??? thx! m.q -------------- next part -------------- An HTML attachment was scrubbed... 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